Noise cancellation apparatus

ABSTRACT

Disclosed is a transducer for use in a noise cancellation apparatus for reducing background noise. The transducer includes a housing having first microphone means for receiving a first acoustic sound composed of speech originating from an operator operating the apparatus and background noise, and for converting the first acoustic sound to a first signal, and second microphone means arranged at a predetermined angle φ in close proximity with respect to the first microphone means for receiving a second acoustic sound composed of substantially the background noise and for converting the second acoustic sound to a second signal. The first and second microphones are connected to a differential amplifier means of the noise cancellation apparatus so as to obtain a signal representing substantially speech. The amplifier means is for receiving acoustic sounds from each microphone and has a first terminal and a second terminal, wherein the second terminal is grounded. The transducer further includes a transistor means for receiving and amplifying an AC signal representative of the audio input from each microphone; and means for filtering the amplified AC signal from the DC signal, so that the DC signal powers the amplifier means. Also disclosed is a method for calibrating an active noise reduction apparatus including a housing having a speaker to produce an acoustic anti-noise signal in the housing, a microphone to detect an external noise signal, and an amplitude adjustment means to calibrate the acoustic anti-noise signal to create a quiet zone in the housing for operation with an independent electrical assembly, wherein the apparatus is calibrated separately from the electrical assembly. The method includes the steps of: inputting the external noise signal received by the microphone to produce an anti-noise signal; transmitting to the speaker the anti-noise signal having an equal gain and opposite phase response to the external noise signal detected by the microphone; and balancing the gain and phase response of the anti-noise signal by the amplitude adjustment means located in the noise reduction apparatus to match the gain and phase response of the external noise signal to yield a theoretical zero in the quiet zone.

RELATED APPLICATIONS

This application is a division of 08/912,469 filed Aug. 18, 1997 whichis a Division of 08/485,047 filed Jun. 7, 1995 now U.S. Pat No.5,732,143 continuation-in-part of U.S. Ser. No. 08/339,126, filed Nov.14, 1994 now U.S. Pat. No. 5,673,325, which is a continuation-in-part ofU.S. application Ser. No. 968,180 filed Oct. 29, 1992 now U.S. Pat. No.5,381,473 issued Jan. 10, 1995, incorporated herein by reference.Reference is also made to U.S. Pat. No. 5,251,263, issued Oct. 5, 1993and incorporated herein by reference.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention relates to method and apparatus for noisecanceling and noise reducing by attenuating unwanted ambient noise fromreaching the eardrum and canceling background acoustic noise receivedfrom a boom microphone or directional microphone, when used with aheadset or boom headset or the like.

The invention further relates to an active noise reduction system foruse in headsets, particularly in the earphone vicinity where the systemutilizes a sensor microphone to detect unwanted, background noise. Thisnoise signal outputted by the sensor microphones is processed byelectro-acoustical means to produce an inverted signal so that a quietzone is created in an acoustical waveguide located between the outputtransducer, and the eardrum. Therefore the desired original audio signalis not disturbed by noise when transmitted to the ear of the user. Theacoustical waveguide absorbs any sound returning to the microphone fromthe ear (preventing feedback) and deadens any sound returning from themicrophone to the ear.

This invention also relates to a noise cancellation apparatus, for usewith a telephone handset or a boom microphone or directional microphonesor the like, where the system utilizes two microphones, a firstmicrophone for receiving sound comprised of speech and background noise,and a second microphone for receiving sound comprised of substantiallybackground noise, with the means for subtracting the second signal fromthe first signal.

The microphone in the noise cancellation system of the present inventionutilizes a two terminal system, in which the output audio signalcomprised of speech and the power support input used to drive the systemare transmitted on one terminal and the second terminal is grounded.

The noise cancellation apparatus of the present invention also relatesto a directional microphone used in a far-field microphone device havingthe ability to accept acoustical sounds in certain directions betterthan in other directions.

The noise cancellation and noise reduction system of the presentinvention may be enhanced by the inclusion of an automatic audiomicrophone transmission feature, a sidetone feature to transmit aportion of the signal to the earcup of the speaker, and a feature toconvert an active noise cancellation microphone to a standardomni-directional microphone by removing voice microphone from thecircuit, and the increasing the gain of the noise microphone amplifier.This enhancement allows all audio from external surroundings to betransmitted to the earcup of the speaker by increasing the sidetonechannel gain without the addition of any other microphone elements.

2. Description of the Prior Art

As is to be appreciated, in numerous situations, the presence ofbackground acoustic noise is undesirable. As an example, consider thesituation in which an operator is attempting to conduct a telephoneconversation from a telephone or such similar device located in a noisyarea. In this situation, loud acoustic background noise is received by amicrophone in the handset of the telephone and converted to anelectrical signal which is supplied to the telephone(s) of the person(s)having the conversation with the operator and is converted thereat to anacoustic signal. As a result, the person to whom the operator iscommunicating constantly hears the loud background noise. Further, whenthe person is speaking, such speech is combined with the backgroundnoise and, as such, may be difficult for the other person(s) tounderstand. As a result, the operator may have to shout into themicrophone of the telephone. Furthermore, the signal representing thebackground noise is also supplied from the microphone in the operator'shandset to the speaker in the operator's handset as sidetone. Thus, theoperator also constantly hears the background noise from the speaker inthe operator's handset and, when the other person is speaking, mayimpair the understanding thereof.

As another example, consider the situation in which a pilot who isoperating a helicopter or the like wishes to communicate with anotherperson by way of radio frequency (RF) communication. In this situation,the pilot typically speaks into a so-called boom microphone or boomheadset which is coupled to a radio transmitting/receiving devicewhereupon the speech is converted into RF signals which are transmittedto a second receiving/transmitting device and converted therein tospeech so as to be heard by the other person(s). As with the abovesituation of a telephone located in a noisy area, the loud backgroundnoise from the helicopter is received and converted into an electricalsignal by the boom microphone or headset device and thereafter suppliedto the receiving device. As a result, the person(s) communicating withthe pilot hears the loud background noise. This may be particularlyannoying when the pilot leaves the radio transmitting/receiving devicein the "ON", (the hot mike) position while operating the helicopter.

As yet another example, consider voice verification and/or recognitionsystems into which an operator must speak for access, for instance to aphysical facility or, to operate a computer or automatic teller machine.Background noise can prevent access (no recognition or verification dueto background noise) or can provide false access by false verification.

In an attempt to reduce background noise so as to improve performance ofa telephone or a boom microphone or headset or the like located in anoisy environment or the like, pressure gradient microphones may beutilized. Basically, a pressure gradient microphone responds to thedifference in pressure at two closely spaced points. When used in anenvironment where the pressure gradient of the background noise isisotropic, the electrical signal produced by the pressure-gradientmicrophone due to such background noise is effectively zero. However, inmost actual situations, the pressure gradient of the background noise isnot isotropic and, as a result, in these situations, the performance ofthe pressure-gradient microphone is adversely affected. Additionally,since voice or speech propagates in more than one direction, theelectrical signal produced by the microphone which corresponds theretois often degraded. Thus, even if a pressure gradient microphone isutilized in either a telephone handset or a boom microphone, the desiredamount of background noise cancellation may not be sufficient and theperformance may not be adequate.

Furthermore, since two opposite sides of a pressure-gradient microphonerespond to acoustic pressure, as previously mentioned, the handset of anexisting telephone would have to be substantially modified so as toenable these two sides of the microphone to respond to the acousticpressure. Moreover, as a result of using such a microphone in atelephone handset, the electrical signals produced therefrom should beamplified. Thus, to replace the conventional microphone in a telephonehandset of an existing telephone with a pressure-gradient microphonewould typically necessitate replacing the handset with a new handsetand, as such, would be relatively expensive.

As an alternative to using pressure-gradient microphones, an acousticfeed-back type system may be utilized. Such a system normally includescompensation filters which are used to equalize the transfer function ofthe output transducers. Since the characteristics of the speakers aretightly controlled by these filters, the cost of the filters isrelatively high. As a result, such acoustic feed-back systems aretypically relatively expensive.

Many microphones used with noise cancellation and noise reductionapparatus are inherently nondirectional or omnidirectional, such as theelectrostatic, piezoelectric, magnetic and carbon microphones. Withomnidirectional small microphones, at low frequencies there issufficient diffraction of sound around the microphone so that diaphragmmotion is insensitive to the direction of the sound. At highfrequencies, and correspondingly shorter wavelengths, the microphonebecomes acoustically larger and shows a preference for sound arrivingperpendicular to the diaphragm. Thus, the smaller in size of themicrophone, the higher in frequency its behavior remainsomnidirectional. Hence, the omnidirectional microphones are smallcompared to the wavelength and the microphone case shields the rear sideof the diaphragm from receiving certain sound waves at different angles.As a result, these prior art microphones are referred to as pressuremicrophones since pressure is a scaler, and not a vector quantity. Thus,a directional microphone response able to increase the sensitivity ofsound in a far-field region from a variety of directions is desired fora microphone device in an active noise cancellation system. That is, toachieve a directional microphone response by adding the outputs of theomnidirectional pattern and bidirectional or "figure-eight" pattern, andthen simply adjusting the amplitude and phase of the summed outputsignal to produce the desired pattern. The figure-eight pattern is alsoknown as a cosine pattern and is mathematically expressed a p=COS θ, inpolar coordinates. In directional microphones, distance is a factor. Thedistance factor measures how much farther away from a source adirectional microphone may be used, relative to an omnidirectionalpattern, and still preserve the same ratio of direct to reverberantpickup. Thus, the prior art has failed to provide a directionalmicrophone in an active noise reduction apparatus based on theomni-directional patterns and the cardioid patterns where the soundpressures arriving at a determined point are added vectorially.

In devising the circuitry for an active noise cancellation apparatus foruse with a boom microphone device or a directional microphone devicecomprising at least two microphones, it is known to use a three terminalmicrophone configuration. That is, a noise cancellation system havingtwo or more microphones connected to an amplifier, for example, requirescircuitry having three terminals: a power supply input terminal, anaudio signal output terminal, and a ground terminal. In an effort toreduce the complexity and cost of the noise cancellation system utilizedin the microphone, or boom microphone or the like which optionally maybe used with a headset of the noise reduction apparatus, a two terminalmicrophone configuration is desired. It is desired to have a microphoneconfiguration where the DC voltage supplied from a power supply isinputted on the same terminal as the AC audio signal outputted from themicrophones, whereby the AC signal is superimposed on the DC signal.Thus, the prior art has failed to provide a two terminal microphoneconfiguration for use in an active noise cancellation apparatus, wherethe power and signal are superimposed on the first terminal and thesecond terminal is grounded

In yet a further attempt to reduce background noise so as to improve theintelligibility of electro-acoustic communication using headsets with amicrophone, a technique has been developed, called active noisereduction that utilizes a sensor microphone placed between the speakerand the ear in the sound field of the speaker, and which senses thebackground noise and programs audio. With this active type headphonedevice, a negative feedback loop is used whereby the electrical signalsconverted from the external noises by a microphone unit are fed back ina reverse phase for reducing the noise in the vicinity of the headphoneunit. A feedback circuit utilizing a closed loop system as shown in theprior art provides a "quiet zone" between the speaker and the ear whicheliminates the background noise. This is because in a noisy environment,the ear will detect not only the output of the speaker, but also thebackground noise.

Reference is made to the following documents providing a closed loopactive noise reduction system, which documents are hereby incorporatedby reference:

U.S. Pat. No. 2,972,018 to Hawley et al.

U.S. Pat. No. 3,098,121 to Wadsworth

U.S. Pat. No. 4,833,719 to Carme et al.

U.S. Pat. No. 5,138,664 to Kimura et al.

Japanese Patent Abstract No. 3-169199 to Saeki.

The above-referenced patents illustrate a variety of noise cancelingdevices. For instance, Hawley et al. relates to a noise reduction systemfor earphones having a plastic casing located between the speaker andthe microphone; Wadsworth provides an earphone having a microphonelocated on top of the headband; Carme et al. is directed to an earphonehaving a hollow annular part located between the speaker and themicrophone; Kimura et al. calls for a noise reduction headphone having acup member located between a speaker and a microphone; and Saeki relatesto a noise canceling headphone having a microphone located between twooppositely facing loudspeakers.

However, there exist various disadvantages in the conventional activenoise reduction systems. The prior active noise cancellation systems,for instance, utilize closed loop-type circuits governed by theassociated equations: ##EQU1## where P=output

S=standard audio signal

H₁ =high pass filter

H₂ =speaker at headset

N=noise component

B=variable gain/phase control

The conventional closed loop noise reduction system is not ideal as avery large direct transmission gain (1+BH1H2) is required in order toreduce the noise component (N) to zero at the output (P). This systemsuffers from the problem of instability. This creates drawback ofoscillation, i.e., squealing due to the unstable loop conditions causedby variations in the transfer function of the speaker, feedbackmicrophone and acoustic cavity containing these elements and userheadgear. The degree of noise cancellation generated by the conventionalclosed loop noise reduction device, at any frequency, is directlyrelated to the direct transmission gain at that frequency. However, thehigher the gain the more susceptible the device is to instability.

The conventional active noise reducing headphone device also has thedrawback that when mechanical vibrations such as impact, frictionalinduced vibrations from connecting cords, user jaw movement inducedvibrations etc., are transmitted to the noise feedback microphone, thesevibrational noises are converted to electrical signals by themicrophone. These signals are amplified and cause instability and othernon-linear effects, for example, audio interruption, loud noises orpressure surges.

Another drawback of conventional active noise reducing headphone devicesis the complexity added to the device to avoid canceling the desiredaudio signal, which signal is inputted as an electrical signal. Thedesired audio signal (S) of the conventional device is input into twosumming nodes to create the signal transmitted to the user's ear. Thefirst summing node adds the negative feedback microphone signal to thedesired input audio signal. But, in a conventional closed loop feedbackdevice, the signal feedback from the microphone contains the desiredaudio signal as well as the ambient noise signal which is desired to becanceled. This feedback signal is subtracted from the desired inputaudio signal to create the anti-noise signal, with zero desired audiosignal content. Then, a second summing node is used to add the desiredaudio signal back into the loop so it can be transmitted to the outputtransducer. This method 5 of generating the desired audio signal addscomplexity and cost to the conventional noise reducing device. Theadditional summing node processing in the conventional device alsoincreases chances of creating distortion in the desired audio signal aswell as increasing the possibility of instability.

In addition, various other prior art headphone configurations have beendeveloped for creating an active noise reduction device, where the inputand output transducers are positioned in relation to the ear, such asthe following three documents, which are incorporated by reference:

U.S. Pat. No. 5,134,659 to Moseley.

U.S. Pat. No. 5,117,461 to Moseley.

U.S. Pat. No. 5,001,763 to Moseley.

Moseley ('659) relates to a noise canceling system for headphones havinga baffle, two speakers, and two microphones wherein the baffle serves toimpede noise from traveling directly from a noise source to the inputtransducer by forcing the noise to travel a longer distance around thebaffle and through a foam barrier. Moseley ('461) is directed to anelectroacoustic function including noise cancellation for use withheadbands having a microphone mounted on the headband to face in samedirection of the ear canal. Moseley ('763) relates to a noisecancellation system for headbands having a speaker, microphone, and abaffle.

Thus, in general, the Moseley patents are concerned with the location ofthe speaker, being the output transducer, and the microphone, which isinput transducer. In fact, the patents require that the speaker andmicrophone be in the same plane or substantially aligned in the sameplane. Also, the patents teach that the processed signal output issubstantially in the same time domain as the original acoustic wave,that is the signal is in phase.

In contrast to the Moseley patents, the present invention is not per seconcerned with the alignment of the speaker and microphone in the sameplane (although such alignment need not be explicitly excluded). Theoutput transducer and microphone utilized in the open loop active noisereduction of the present invention may be perpendicular, tangential, orin any other location out of the same plane (as well as in the sameplane). The present invention provides a noise reduction system havingthe capability to transmit the original input audio signal to thespeaker without the readdition of the input audio signal. This isbecause the sensor microphone, which is the control action of the openloop, is so disposed from the audio signal, that the audio signal is notdetected by the pickup or sensor microphone. That is, in the open loopsystem of the present invention, the original desired audio signal istransmitted to the speaker independent of the ambient noise detected bythe microphone. In addition, in the present invention an acousticalmaterial can be located between the output transducer and the eardrum ofthe user to create an acoustical waveguide for the transducer bycoupling the audio signal to the ear of the user. The acousticalmaterial located between the output transducer and microphone acts as anacoustic filter to decrease the open loop gain by placing an acousticalimpediment in the path of the pickup microphone and the outputtransducer. The acoustical material isolates the desired originalinputted audio signal from the noise detected and canceled by the pickupmicrophone. The background noise signal detected by the pick-upmicrophone is inverted through electric-acoustical processing meansproducing an anti-noise signal, which signal is transmitted to theacoustical waveguide to create a quiet zone. This quiet zone is locatedbetween the output transducer and the eardrum of the user.

Thus, the prior art has failed to provide a relatively low-cost meansfor reducing background noise to an acceptable level for use withcommunication systems or the like, and a cost-effective means forenabling existing audio communication systems to reduce background noiseto an acceptable level.

OBJECTS AND SUMMARY OF THE INVENTION

An object of the present invention is to provide an active noisecancellation apparatus and an active noise reduction apparatus to createa noise reducing system which overcomes the problems associated with theprior art.

More specifically, it is an object of the present invention to providean active noise cancellation apparatus and active noise reductionapparatus which reduce background noise to an acceptable level.

Another object of the present invention is to provide noise reductionapparatus for use with a headset device and boom microphone or toprovide a noise cancellation microphone device or the like.

It is still another object of the present invention to provide noisereduction and cancellation apparatus and an active noise reducing systemas aforementioned which is relatively inexpensive.

It is yet another object of the present invention to provide arelatively low-cost noise reduction and cancellation apparatus for usewith telecommunication systems which is operable with standard availableon-line power.

Another object of the present invention is to provide an enhanced activenoise cancellation and noise reduction headset by adding a talk thrufeature, which enables the user to hear the microphone audio signals aswell as the external audio from the surrounding environment, without thephysical addition of any other microphone elements. The object of thepresent invention is to have an active noise cancellation and noisereduction headset where all the audio from external area is transmittedto the earcup of the speakers by increasing the gain of the sidetonechannel. This active noise cancellation microphone of the presentinvention is converted to a standard omni-directional microphone byremoving the voice microphone from the electronics and increasing thegain of the noise microphone amplifier.

A still further object of the present invention is to provide arelatively low-cost noise cancellation apparatus which is readilyadaptable to handsets of existing communication systems and which isoperable with standard available on-line power.

A yet further object of the present invention is to provide a relativelylow-cost noise reduction apparatus for use with audio communicationsystems which enables the user to selectively amplify a received signalor, which may be used in a boom microphone with a headset or, which maybe used as a noise canceling microphone.

In many applications as described herein, microphones withother-than-omnidirectional characteristics are desired. Such microphonesreject signals from certain directions and thus yield an improvement ofthe signal-to-noise ratio. The directional microphones based onsummation scheme, which is that of the present invention, may depend onthe algebraic combinations of the sound pressure signals with phasedifferences which are exclusively due to the electronics of the system.As opposed to gradient-type microphones, the directivity of suchmicrophones is dependent on the ratio of linear dimensions towavelength.

When two or more microphones are fed into the same amplifier, it ispossible that signals from a sound source at distance from themicrophones may arrive at the microphones 180° out of phase, cancelingeach other. Therefore, it is an object of the present invention toensure the omni-directional and directional microphones are phasedproperly.

It is also an object of this invention that the first and secondmicrophones arranged at a predetermined angle and/or distance withsubtraction apparatus disclosed herein can also be used in the area ofambient noise cancellation for microphones in acoustic surveillance ortelemetry or even directional microphones such as directionalmicrophones with sidelobes.

Accordingly, is an object of the present invention to provide a low costmicrophone for use in a noise cancellation system withother-than-omnidirectional characteristics.

It is a further object of the present invention to provide acontrollable variety of directivity patterns with a microphone based onthe magnitude and phase lobe construction.

It is yet another object of the present invention to provide adirectional microphone by adding vectorially at a determined point thesound pressures arriving at that point from all simple sources.

It is still another object of the present invention to provide atwo-terminal microphone system, including the directional microphone asaforementioned, in an active noise cancellation environment, whichallows the audio output signal to be superimposed on the voltage inputsignal at the same terminal.

Another object of the invention to provide a novel active noisereduction apparatus for use in headsets due to its simplicity and lowcost circuitry by positioning elements in an open loop system.

It is object of the present invention to provide a noise reductionapparatus in which the ambient noise is attenuated in a regular mannerwithout being degraded by mechanical or vibration induced microphonesignals.

It is another object of the present invention to provide an active noisereducing system comprised of a headset, handset or the like with a boommicrophone or directional microphone or the like which isunconditionally stable due to its open loop configuration.

It is further object of the present invention to reduce the powerrequired by the noise reduction apparatus by coupling theelectro-acoustic transducer efficiency.

It is further object of the present invention to reduce the complexityand/or cost of the active noise reduction circuit by employing a methodof combining the desired audio signal and the anti-noise signal to theoutput transducer in a single summing node.

It is further object of the present invention in a noise reducing systemto reduce anti-noise processing induced distortion of the desiredelectrical input signal which is converted to an acoustic signal andtransmitted to the ear in a noise reduction system.

Another object of the noise reduction apparatus involves a sensor orpickup microphone placed behind or in front of the output transducer,and outside of the sound field and the plane of the speaker, so that themicrophone detects only the background noise by utilizing of theacoustical material, which performs dual functions.

It is a further object of this invention to provide an acousticalmaterial as an acoustic filter when positioned over a microphone, and asan acoustic waveguide when placed between the output transducer and earof the user.

It is the microphone that is the control action of the system, themicrophone is independent of the inputted audio signal, the desiredoutput. A resilient acoustical waveguide is preferably positionedbetween the speaker/microphone and the ear to create a quiet zone. Thiswaveguide is preferably more than just the usual rubber sponge which iscommonly provided on earphones for comfort purposes. One type of suchmaterial is called "Slo-Flo" foam and it is of such a density andconstruction so as to define a noise-free response and to deaden anysound reflections returning to the microphone, acting as an acousticalfilter, from the listener's face and/or ear; whereas the prior art usesa negative feedback of the signal from the microphone, no such feedbackis produced in the present invention. Instead, an open-loop arrangementis utilized, wherein there is no need to add another audio signal, butthe original input audio signal is transmitted to the speaker, as thesignal has not been disturbed by the open loop system.

It is important to understand the distinctions between a conventionalclosed-loop reduction apparatus and the novel open loop reductionapparatus of the present invention.

An open loop system of the present invention is one in which the controlaction is independent of the output or desired result. A closed loopsystem is one in which the control action is dependent on the output.The key term in these definitions is control action. Basically, the termrefers to the actuating signal of the system, which in turn representsthe quantity responsible for activating the system to produce a desiredoutput. In the case of the open loop system, the input command is thesole factor for providing the control action, whereas for a closed loopsystem, the control action is provided by the difference between theinput command and the corresponding output.

To complete the comparison of the closed loop versus open loopoperation, certain performance characteristics of each system is asfollows: open loop systems have two outstanding features, namely, theability to perform a function being determined by calibration andsimplicity in construction, for instance because the problems ofinstability are not incurred. For closed loop systems, a noteworthyfeature is the ability to faithfully reproduce the input owing to thefeedback, since the actuating signal is a function of the deviation ofthe output from the input; this control action forces the actuatingsignal almost at zero. A major disadvantage of this feedback factor isthat it is responsible for one of the greatest difficulties in using aclosed loop systems, namely the tendency to oscillate.

The active noise reduction apparatus as well as the noise cancellationapparatus can be used in any telecommunication systems that are used inflight (e.g., helicopter or airplane) or in other settings such astelephones, or voice recognition and/or verification systems forinstance, for access to a physical facility or to a computer (either viadirect or indirect interface or via telephone lines) or to an automaticteller machine or, in other recognition and/or verification systems.

The noise cancellation apparatus comprises: a housing having firstmicrophone means for receiving a first acoustic sound composed of speechoriginating from an operator operating said apparatus and backgroundnoise, and for converting said first acoustic sound to a first signal,and second microphone means arranged at a predetermined angle .Oslashed. in close proximity with respect to said first microphone meansfor receiving a second acoustic sound composed of substantially saidbackground noise and for converting said second acoustic sound to asecond signal; and means for subtracting the second signal from thefirst signal so as to obtain a signal representing substantially saidspeech. The two terminal transducer for use in the noise cancellationapparatus for reducing background noise comprises: a plurality ofmicrophones connected to an amplifier means of the noise cancellationapparatus; the amplifier means for receiving audio signals from themicrophone having a first terminal and a second terminal wherein thesecond terminal is grounded; a voltage means inputting a DC signal onthe first terminal; a transistor means connected to the first terminalfor receiving an AC signal from the microphones; means for superimposingthe AC signal onto the DC signal on the first terminal; means forfiltering the AC signal from the DC signal, so the DC signal powers theamplifier means; and means for outputting the AC signal generated by themicrophones at the first terminal. The directional microphone for use inobtaining a far-field response when speaking into a boom microphone ofan active noise cancellation apparatus, which accepts sounds in avariety of directivity patterns comprises: a housing having an array ofspaced microphones means for receiving acoustics signals and outputtingelectrical signals having a spaced separation between the microphones; apressure sound source inputted into the housing as a sinusoidal soundwave having a magnitude and phase which intersects the microphones at apredetermined distance to form an angle; means for calculating thedistance from each microphone to the sound source; a summing channel foradding the output signals of the array of microphones to obtain a sumoutput signal; a signal processing means to produce an acoustic signalrepresenting only speech from the sum output signal; and means foradjusting the magnitude of the sum signal to produce the desiredresponse pattern. The open-loop active noise reduction apparatus forreducing ambient noise in the vicinity of the eardrum comprises; ahousing for receiving an input and to signal; an output transducerlocated in the housing; an input transducer for detecting and reducingambient noise located not in substantially the same plane as the outputtransducer; an open-loop signal processing means to reduce the ambientnoise detected by the input transducer; means for transmitting the inputaudio signal to the eardrum without disturbance of the ambient noise; anacoustic means to isolate the output transducer from the inputtransducer for channeling the input audio signal representingsubstantial speech between the output transducer and the eardrum,wherein a quiet zone is created to isolate sound transmitted from theinput transducer. The open loop noise reduction system for use with anactive noise cancellation apparatus comprises; a pick-up microphone fordetecting noise signals to convert to electrical signals; a speakerlocated in the headset having a acoustic means; an audio transmissionsignal; means for electrically rejecting vibrations of the electricalsignal; a variable gain/control means for adjusting the noise signal;means for filtering out mechanical vibration induced low frequencydisturbance from reaching the speaker; a summing node to combine theanti-noise signal and the noise signal to produce a quiet zone in theacoustic means; means for transmitting the audio signal to the speaker;and means for maintaining phase agreement between the noise signal andthe anti-noise signal of the speaker.

An open loop active noise reduction apparatus for reducing ambient noisein the vicinity of the eardrum, comprising: a housing for receiving aninput audio signal; an output transducer located in the housing; aninput transducer for detecting and reducing ambient noise located in thehousing; a open loop signal processing means to reduce ambient noisedetected by the input transducer; means for transmitting the input audiosignal without disturbance of the noise to the eardrum; an acousticmeans to isolate the output transducer from the input transducer forchanneling the input audio signal representing substantial speechbetween the output transducer and the eardrum and creating a quiet zoneto isolate sound transmitted from the input transducer.

An active noise cancellation and noise reduction system for use in aheadset for transmitting audio signals from microphones and forreceiving external audio from a surrounding environment comprising: afirst microphone means having a first switch means having a noisecanceling mode and a first talk thru mode and a second microphone means;a microphone amplifier means connected to the microphones by a secondswitch means; an audio microphone transmission means for connecting theamplifier means when the first switch means and the second switch meansare operating in the noise canceling mode, wherein the microphonetransmission means is bypassed when the first switch means and thesecond switch means are in a talk-thru mode; a transmission gate fortransmitting the audio signal from both microphone means to a bufferamplifier when the first switch means and second switch means are in thenoise canceling mode, wherein the transmission gate is disabled when thesecond switch means is in the talk thru mode; the buffer amplifier meansfor transmitting the audio signal received from the transmission gate toan audio system and to a scaling amplifier when the first switch meansand second switch means are operating in the noise canceling mode;

the buffer amplifier means for directly outputting the audio signalsreceived by the microphone amplifier means when the first switch meansand second switch means are operating in the talk thru mode; the scalingamplifier having a third switch means having a noise canceling mode anda talk-thru mode, provides a sidetone signal to an earcup of a speakertransmitted from both microphone means and from the external audio whenall the switch means are operating in the noise canceling mode; thescaling amplifier having a gain control, wherein the gain control isincreased when all the switch means are operating in the talk thru modeto increase the sidetone signal to a speaker; active noise reductionsystem receives and outputs the sidetone signal to the speaker in theheadset.

Other objects, features and advantages according to the presentinvention will become apparent from the following detailed descriptionof the illustrated embodiments when read in conjunction with theaccompanying drawings in which corresponding components are identifiedby the same reference numerals.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 illustrates a telephone having a noise cancellation apparatusaccording to an embodiment of the present invention;

FIG. 2 is a block diagram of the noise cancellation apparatus used inthe telephone of FIG. 1;

FIG. 3A is a front plan view of the receiver portion of the telephone ofFIG. 1;

FIG. 3B is a side elevational view of the receiver portion of thetelephone of FIG. 1 with the top removed;

FIG. 4 is a schematic diagram of the block diagram of FIG. 2;

FIG. 5 is another schematic diagram of the noise cancellation apparatusillustrated in FIG. 2;

FIGS. 6A, 6B and 6C illustrate a boom microphone device utilizing anoise reduction apparatus according to an embodiment of the presentinvention;

FIGS. 7A and 7B are schematic diagrams to which reference will be madein explaining the operation of the present invention;

FIG. 8 illustrates a noise reduction apparatus according to the presentinvention;

FIGS. 9A, 9B, 9C, 9D, 9E and 9F illustrate boom microphone and headsetembodiments of the present invention (FIGS. 9A and 9B each showing anembodiment having particular placement of the microphone; FIG. 9Cshowing an overview of the headset, having active noise reductionapparatus FIGS. 9D, 9E, and 9F showing side views of the boommicrophone) and FIG. 9F shows a preferred embodiment of the active noisereduction apparatus utilized in a headset;

FIGS. 10A and 10B are schematic diagrams of the noise reductionapparatus of FIG. 8;

FIG. 11 illustrates a phase reversing circuit;

FIG. 12 illustrates an oppositely charged microphone circuit;

FIGS. 13A and 13B illustrate active cancellation curves from embodimentsof the invention;

FIG. 14 is a schematic diagram of a directional microphone.

FIG. 15 is a schematic diagram of a linear array of microphones to whichreference will be made in explaining the operation of an embodiment ofthe present invention;

FIGS. 16A and 16B illustrate an alternative embodiment of the schematicdiagram of an array of microphones in a cylinder and the schematicdiagram of the electrical circuit array;

FIGS. 17A and 17B illustrate block diagrams of the prior art threeterminal microphone configuration and a two terminal microphoneconfiguration of the present invention;

FIG. 18 is a schematic diagram of a simple two terminal microphonecircuit;

FIG. 19 illustrates a preferred embodiment of a two terminal microphonecircuit used in an active noise cancellation system;

FIG. 20 is an alternative embodiment of a two terminal microphonecircuit used in an active noise cancellation system;

FIG. 21 is a schematic diagram of the prior art closed loop active noisereduction system;

FIG. 22 is a schematic diagram of a open loop active noise reductionsystem of the present invention;

FIGS. 23A and 23B illustrate a perspective view and side view of theacoustical waveguide of the present invention;

FIG. 24 is a diagram of the quiet zone achieved by utilizing the activenoise reduction system;

FIG. 25 illustrates a preferred embodiment of an active noise reductionsystem of the present invention;

FIG. 26 illustrates another preferred embodiment of an active noisereduction system of the present invention;

FIG. 27 illustration alternative embodiment of the active noisereduction system of the present invention including a high pass filter;and

FIG. 28 is a block diagram of the active noise reduction and noisecancellation system utilized in a headset having the talk thru,sidetone, and automatic audio microphone transmission features.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS

FIG. 1 illustrates a telephone 8 which utilizes a noise reductionapparatus in accordance with an embodiment of the present invention. Asshown therein, the telephone 8 generally includes a handset 10, having aspeaker portion 41 and a receiver portion 42, and a telephone unit 18which may be coupled therebetween by way of a telephone cord 30.Alternatively, the telephone may be a cordless type telephone and, assuch, the handset 10 is coupled to the telephone unit 18 by way of RFwaves. The receiver portion 42 includes first and second microphones 12and 14, respectively, (FIG. 2), a switch 40 for adjusting the volume ofa signal supplied to the speaker portion 41, and a cap 48 having arecessed portion 44 and a mesh portion 46.

FIG. 2 illustrates the telephone 8 in block diagram form. As showntherein, the handset 10 generally includes first and second microphones12 and 14, respectively, a subtracting device 16, which in a preferredembodiment is an operational amplifier ("op-amp"), an amplifier 20,which is preferably an op-amp, and a speaker 22. The first and secondmicrophones 12 and 14, respectively, op-amp 16 and amplifier 20 arepreferably contained within the receiver portion 42 (see FIG. 1).

Acoustic signals composed of speech or the like and background noise aresupplied to the first microphone 12 and converted therein into acorresponding electrical signal which is thereafter supplied to the plusterminal of the op-amp 16. The background noise is supplied to thesecond microphone 14 and converted therein into a correspondingelectrical signal which is thereafter supplied to the minus terminal ofthe op-amp 16. The op-amp 16 is adapted to subtract the noise signalfrom the second microphone 14 from the speech and noise signal from thefirst microphone 12 and to supply therefrom an electrical signalrepresenting substantially the speech to the telephone unit 18 whereuponthe speech signal is transmitted therefrom through the telephone linesto a desired telephone or telephones. The output signal from the op-amp16 is also combined in the telephone unit 18 with a received signal fromthe telephone lines and supplied to the amplifier 20. The op-amps 16 and17 are preferably relatively low-power integrated circuits (IC's), suchas complementary metal oxide semiconductors (CMOS), and may beconstructed from either one or more CMOS IC chips. Although not shown inFIG. 2, amplifier 20 may be selectively set by use of the switch 40(FIG. 1) by the operator so as to adjust the amplification of thereceived signal to a desired level. The amplified signal from theamplifier 20 is supplied to the speaker 22, whereupon the amplifiedsignal is converted into an acoustic signal so as to be heard by theoperator.

FIGS. 3A and 3B illustrate two views of the receiving portion 42, inwhich the cap 48 is removed in the view of FIG. 3A. As shown therein,the receiving portion 42 generally includes a housing 74, a circuitboard assembly 78, the first and second microphones 12 and 14,respectively, and the cap 48. The first and second microphones 12 and14, respectively, which are preferably electric microphones or similarsuch microphones, are arranged or positioned as hereinafter described.These microphones are held in place or secured by a holding member 76which, for example, may be constructed of a foam-like material, which,in turn, is secured to the housing 74. The respective outputs from thefirst and second microphones 12 and 14 are supplied through respectivewires (not shown) to the op-amp 16 which is contained on the circuitboard assembly 78 which, in turn, is attached to the housing 74. Ashereinafter more fully described, the circuit board 78 may containadditional circuit elements for processing the signals received from thefirst and second microphones and for amplifying signals for supply tothe speaker 22 (FIG. 2). A cover 72 may be utilized which is attached tothe housing 74 by use of adhesives or the like or alternatively may besonically welded together. The cover 72 and the housing 74 with thecircuit board assembly 78, holding member 76 and the first and secondmicrophones 12 and 14 form an assembly 71.

The cap 48, which may be constructed from a plastic-type material suchas polycarbonate, includes an annular side member 43 and a portion 45having a typical thickness T which is coupled to the side member 43 andarranged so as to be lower than the upper portion of the side member bya minimum predetermined amount such as 0.020 of an inch, therebycreating a recessed portion 44. The portion 45 includes a portion 46having a thickness T' which is less than the thickness T and which has aplurality of through holes contained therein and may resemble amesh-like portion. In a preferred embodiment, the thickness T', of theportion 46 has a thickness of less than 0.030 of an inch. Since theportion 46 represents a relatively small amount of the portion 45,reducing the thickness therein does not adversely affect the overallstructural rigidity of the cap 48. Alternatively, the portion 46 may beconstructed from a stronger material, for example, stainless steel orsuch similar material, and combined with the portion 45. As is to beappreciated, by arranging the portions 45 and 46 so as to be recessedfrom the upper portion of the side member 43, even when the receiverportion 42 is placed on a surface, the side member 43, and not theportions 45 or 46, contact such surface. As a result, any loads are notdirectly impacted on the portion 45 and/or the portion 46, but areinstead delivered to the side member 43.

The cap 48 is positioned over the assembly 71 so that the first andsecond microphones 12 and 14, respectively, are arranged below theportion 46 with the first microphone positioned relatively close to theunderside of the portion 46. Thus, the speech travels a relatively shortdistance from an operator, who is speaking into the receiver portion 42from a distance of preferably less than 1 inch, through the portion 46to the first microphone. As a result, acoustic distortions areminimized.

The arrangement of the first and second microphones 12 and 14,respectively, within the receiver portion 42 is illustrated in FIGS. 3Aand 3B. More specifically, as shown in FIG. 3B, the first and secondmicrophones are arranged so as to have an angle .O slashed.therebetween, which preferably has a value in a range between 30° and60°. The first and second microphones are further respectively arrangedso as to have an angle θ and [(90-θ)+.O slashed.] between a planeparallel to the receiving or "sensitive" surface of the first microphone12 and the direction of speech from an operator, and an axis normal tothe sensitive surface of the second microphone 14 and the direction ofspeech, as shown in FIG. 3B; and so as to have an angle ψ between thedirection of speech and the second microphone, as shown in FIG. 3A. In apreferred embodiment, the angle θ has a value of less than approximately35° and the angle ψ has a value of approximately 180°. As a result ofarranging the first and second microphones in this manner, the firstmicrophone 12 receives both the speech from the operator and thebackground acoustic noise which is present in the vicinity, and thesecond microphone 14 essentially receives only the same backgroundacoustic noise which is received by the first microphone.

Although, as previously mentioned, the angle .O slashed. has a valuewhich is preferably between 30° and 60°, the first and secondmicrophones 12 and 14, respectively, may nevertheless operatesatisfactorily even if arranged so as to have an angle .O slashed. whichlies outside this range. However, as the angle .O slashed. becomessubstantially smaller than 30° or larger than 60°, the performance maybe adversely affected. That is, when the angle .O slashed. becomessubstantially smaller than 30°, the second microphone 14 receives boththe speech and background noise. As a result, upon subtracting theoutput signal of the second microphone 14 from the output signal of thefirst microphone 12, a portion or all of the speech may be canceled. Onthe other hand, when the angle .O slashed. is substantially larger than60°, the background noise received by the second microphone 14 may notbe similar to that received by the first microphone 12. As a result,subtracting the output signal of the second microphone 14 from theoutput signal of the first microphone 12 may not adequately cancel thebackground noise received by the first microphone.

In a like manner, although the angles θ and ψ have preferred values ofless than 35° and approximately 180°, respectively, as previouslymentioned, the first and second microphones may operate satisfactorilyeven if arranged so as to have different values of these angles.However, as the values of the angles θ and ψ become substantiallydifferent from the respective preferred values, the performance may beadversely affected. That is, when the angle θ becomes substantiallylarger than 35°, the second microphone 14 may receive both the speechand background noise. Similarly, when the angle ψ is substantiallysmaller or larger than 180°, the second microphone 14 may receive boththe speech and background noise. As a result, in either of thesesituations, upon subtracting the output signal of the second microphone14 from the output signal of the first microphone 12, a portion or evenall of the speech may be canceled.

As is to be appreciated, by using the above-described devices andmaterials for the components of the receiver portion 42, the cost forconstructing such receiver portion is relatively low. Further, by usingCMOS chips, as previously described, the power consumption of thereceiver portion is kept relatively low. As a result, the receiverportion may be powered by the standard power available in the handsetand, as such, does not require additional power or transformers or thelike. Furthermore, although the receiver portion 42 has been describedfor assembly with the handset 10 of the telephone 8, which is a newtelephone, such receiver portion, or a slight variation thereof, may beused in handsets of existing telephones. That is, in this lattersituation, the cap and microphone contained within the handset of anexisting telephone are merely replaced with the receiver portion 42.Thus, such use of the receiver portion 42 provides a relatively easy andlow-cost means to modify a handset of an existing telephone to includethe present noise reduction apparatus.

FIG. 4 illustrates a schematic diagram of one circuit arrangement of thetelephone 8 shown in FIGS. 1 and 2. As shown in FIG. 4, the firstmicrophone 12 is coupled through a resistor 202, which is adapted tofunction as a current limiting resistor so as to correct the bias of anoutput from the first microphone, to an input power terminal 200. Thefirst microphone 12 is further coupled through a resistor 210 to theplus terminal of the op-amp 16 and through a resistor 212 to a variableresistor 214. The second microphone 14 is coupled through a variableresistor 208, which is adapted to function as a current limitingresistor so as to correct the bias of an output of the secondmicrophone, to an input terminal 201, and to the minus terminal of theop-amp 16. The limiting resistor 208 is preferably a variable currentlimiting resistor which enables the level of the output signal from thesecond microphone to be matched to within a predetermined value to thelevel of the output signal of the first microphone 12. Morespecifically, the limiting resistor 208 enables the output signal of thesecond microphone 14 to be weighted such that when a signal having asimilar level is outputted from the first microphone 12, the amplitudeof the difference therebetween is minimized. The value of the currentlimiting resistor 208 can be selected according to minimizationcriteria. A power terminal 198 is connected to resistors 204 and 206,which are adapted to divide the voltage received at the input powerterminal 198, and to the minus terminal of the op-amp 16. The output ofthe op-amp 16 is coupled to capacitors 220, 222 and 226 and resistors224 and 228 which, in turn, is connected to a "microphone input"terminal of the telephone unit 18. The output from the op-amp 16 isfurther coupled through a variable resistor 214, a resistor 216 and acapacitor 218 to ground. Resistors 210, 212 and 216 and variableresistor 214 provide variable gain, for example, 20 to 1 amplification,to the output of the op-amp 16. The capacitors 218, 220 and 222 areadapted to remove residual DC (direct current) levels which may bepresent in the output signal from the op-amp 16. The resistors 224 and228 and the capacitor 226 are adapted to function as a low-pass filterhaving a break point at a predetermined value which, for example, may be3.7 kHz.

The telephone unit 18 is further connected to the telephone lines and isadapted to receive signals through the microphone input terminal and tosupply these signals to the desired telephone or telephones by way ofthe telephone lines. The telephone unit 18 is further adapted to receivesignals from another telephone or telephones by way of the telephonelines and to combine such signals with those received through themicrophone input terminal, as previously described, and to supply thecombined signal to a speaker input terminal 231. The input terminal 231is connected through a capacitor 230, which is adapted to block DCsignals, and a resistor 232 to the minus terminal of an op-amp 17 andthrough a resistor 234 to a variable resistor 240. An power terminal 199is connected to the plus terminal of the op-amp 17. The output from theop-amp 17 is connected through capacitors 242 and 244 and a resistor 246to the speaker 22. The output from the op-amp is further connectedthrough the variable resistor 240, a resistor 238 and a capacitor 236 toground.

The operation of the telephone 8 shown in FIG. 4 will now be describedbelow.

Upon activating the handset 10, by lifting the handset 10 from theswitch hook (not shown) or the like, standard telephone line voltage isapplied to input terminals 198, 199, 200 and 201. A signal from thefirst microphone 12, which has been bias corrected by the currentlimiting resistor 202, is supplied through the resistor 210 to the plusterminal of the op-amp 16. An output signal from the second microphone14, which has been bias corrected by the current limiting resistor 208,is supplied to the minus terminal of the op-amp 16. The op-amp 16subtracts the signal received from the second microphone 14 from thatreceived from the first microphone 12 and outputs the resultingsubtracted signal. DC levels which may be present in the output signalare removed and the signal is amplified. High frequency signals, such asthose over 3.7 kHz, are then removed from the amplified output signaland the resulting signal is supplied to the telephone unit 18. Thus, avoltage signal is supplied to the telephone unit 18 which isproportional to the difference between the voltages generated by thefirst and second microphones 12 and 14, respectively.

An output signal from the telephone unit 18, which is a combination ofthe signals received through the microphone input terminal and thetelephone lines, is supplied to the input terminal 231 of the amplifier17. The signal from the input terminal 231 is supplied to the capacitor230 so as to remove any DC signals which may be present. The output fromthe capacitor 230 is supplied through the resistor 232 to the minusterminal of the op-amp 17. The op-amp 17 buffers the signal from thetelephone unit 18 and supplies the received signal plus the sidetone toop-amp input 231. Such signal may be selectively amplified, through theuse of resistors 232, 234 and 238 and variable resistor 240, by theoperator by use of the switch 40 (FIG. 1). Any DC signals which may bepresent in the amplified signal are thereafter removed by the capacitors242, 244 and 236. The output signal from the capacitor 244 is currentlimited by the resistor 246 and is thereafter supplied to the speaker 22so as to be converted thereat into an acoustic signal.

FIG. 5 illustrates an alternative arrangement for processing the signalsobtained from the first and second microphones 12 and 14, respectively,so as to provide a current output for supply to the telephone unit 18which is proportional to the difference of the voltages generated by thefirst and second microphones.

More specifically, the circuit arrangement of FIG. 5 includes a handset10' having a plurality of input terminals 300, 301, 370 and 390 whichare each adapted to receive standard available on-line power. The firstmicrophone 12 is coupled through a current limiting resistor 302 to theinput power terminal 300 and is further coupled to the plus terminal ofa subtracting device 316, which is preferably a CMOS op-amp. The outputfrom the second microphone 14 is coupled through a variable currentlimiting resister 308 to the input terminal 301 and is further coupledto the minus terminal of the op-amp 316. The signal outputted from theop-amp 316 is supplied through filtering stages 350 to the minusterminal of a subtracting device 351 which is preferably a CMOS op-amp.The filtering stages 350 are adapted to provide a predeterminedfrequency response characteristic such as a signal roll-off at apredetermined frequency. As is to be appreciated, although two filteringstages are shown in FIG. 5 any number of filtering stages may beutilized. The input power terminal 390 is coupled to resistors 392 and394, which are adapted to reduce the signal supplied thereto, and to theplus terminal of the op-amp 351. An output signal from the op-amp 351 issupplied to the base of a transistor 366. The input power terminal 391is connected to a Zener diode 360, a capacitor 362 and a resistor 364which, in turn, is connected to the collector of the transistor 366 andto the microphone input terminal of the telephone unit 18. The emitterof the transistor 366 is coupled through resistors 367 and 368 to theminus terminal of the op-amp 351 so as to provide a feedback loopthereto. The op-amp 351 and the associated components provide electricalisolation between the filtering stages 350 and the transistor 366. Thetransistor 366 is adapted to amplify the signal supplied to thetelephone unit 18.

The output from the telephone unit 18 is coupled to the input terminal231 (FIG. 4) and is thereafter processed in the manner previouslydescribed with reference to the handset 10 of FIG. 4 so as to provide anacoustic signal from the speaker 22.

The operation of the telephone 8' will now be described below.

Upon applying power to the handset 10', by lifting the handset from theswitch hook (not shown) or the like, a portion of telephone line voltageis applied to input terminals 300, 301, 370, 390 and 391. A signal fromthe first microphone 12, which has been bias corrected by the currentlimiting resistor 302, is supplied to the plus terminal of the op-amp316. An output signal from the second microphone 14, which has been biascorrected by the current limiting resistor 308, is supplied to the minusterminal of the op-amp 316. The resistor 308 is preferably a variablycurrent limiting resistor which enables the level of the output signalfrom the second microphone 14 to be matched to within a predeterminedvalue to the level of the output signal of the first microphone 12, in amanner substantially similar to that previously described for resistor208. The output difference signal from the op-amp 316 is provided thoughthe filtering stages 350, which may include one or more RC networks orequivalent circuits, so as to limit the upper frequency of the outputsignal to a predetermined value which, for example, may be 3.7 kHz. Theoutput signal from the filtering stages 350 is supplied to the minusterminal of the op-amp 351 and a voltage signal from the input powerterminal 390, which has been divided to a predetermined value such asone half thereof, is supplied to the plus terminal of the op-amp 351which amplifies the corresponding output signal to the base of thetransistor 366. The voltage from the input power terminal 391 issupplied through the resistor 364 to the collector of the transistor366. As a result, an amplified signal is supplied from the handset 10'to the telephone unit 18 for supply therefrom through the telephonelines to the desired telephone(s) and for combining with a receivedsignal from the telephone(s) for supply to the input terminal 231 in amanner similar to that previously described with reference to FIG. 4.

The individual circuit components without reference designationsdepicted in FIGS. 4 and 5 are connected as shown and will not bediscussed further, since the connections and values are apparent tothose skilled in the art and are not necessary for an understanding ofthe present invention.

FIGS. 6A, 6B and 6C illustrate a boom microphone 100 which utilizes anoise cancellation apparatus in accordance with an embodiment of thepresent invention. More specifically, the boom microphone 100 generallyincludes a housing 174, a circuit board assembly 178, first and secondmicrophones 112 and 114, respectively, and a portion 147. The housing174, which may be constructed from either a plastic-like or metal-typematerial, includes a circular portion 108 having a hole therethrough soas to enable a shaft 106 to be inserted therein. As a result, the boommicrophone 100 may rotate about the shaft 106 as illustrated in FIG. 6A.

The first and second microphones 112 and 114 are respectively coupled tothe circuit board assembly 178 by wires 102 and 104. The circuit boardassembly 178 contains circuitry similar to that on the circuit boardassembly 78 which, as previously described, processes the signals fromthe first and second microphones 12 and 14, respectively, for supply tothe telephone unit 18 and, as such, in the interest of brevity, will notbe further described herein. Therefore, the circuit board assembly 178is adapted to receive a speech and background noise signal from thefirst microphone 112 and to subtract therefrom the background noisesignal from the second microphone 114 so as to derive a signal whichrepresents substantially the speech. Such signal is supplied to atransmitting device (not shown) so as to be converted to a RF signal andtransmitted to a remote receiving device (not shown). The first andsecond microphones 112 and 114, respectively, are held in place by aholding member 176 which, for example, may be constructed of a foam-likematerial. A mesh-like screen 146 which, for example, may be fabricatedfrom a plastic-type or a metal material or the like, is attached to thecut away portion 147 so as to protect the first and second microphones.The mesh 146 has a predetermined thickness which, for example, may beapproximately 0.030 or less of an inch.

The first and second microphones 112 and 114, respectively, which may beelectret microphones, are arranged in a manner similar to that of thepreviously described first and second microphones 12 and 14,respectively, of the handset 10. That is, the first and secondmicrophones 112 and 114, are respectively positioned so as to have anangle θ and [(90-θ')+.O slashed.'] between a plane parallel to thereceiving or sensitive surface of the first microphone and the directionof speech from an operator, and between an axis normal to the sensitivesurface of the second microphone and the direction of speech, as shownin FIG. 5A. Further, the first and second microphones 112 and 114,respectively, are arranged so as to have an angle .O slashed.'therebetween, which has a preferred value in a range between 30° and60°. The first and second microphones 112 and 114, respectively, arelocated in relatively close proximity to the mesh 146 and the cut awayportion 147 of the housing 174 so as not to receive acoustic soundswhich have been unacceptably distorted.

Although the above embodiments have been described as having only onefirst microphone 12 (112) and one second microphone 14 (114), theinvention is not so limited and any number of microphones may beutilized for the first microphone and/or the second microphone. Forexample, a receiver portion 42' (not shown) may be configured whichincludes two or more microphones operating as a first microphone 12'(not shown) and two or more microphones operating as a second microphone14' (not shown). In this configuration, when using multiple microphonesfor the first and/or second microphones, respective variable currentlimiting resistors are preferably provided for all but one microphonefor the first microphone 12' and for all microphones for the secondmicrophone 14'. Thus, the outputs from the first and second microphones,12' and 14', respectively, would comprise a weighted sum of several suchmicrophone output voltages. The current limiting resistors arepreferably set to respective values so as to minimize some functional ofthe difference of the first and second microphones 12' and 14',respectively. The criterion for selecting the values of the currentlimiting resistor or equivalently the weighing function of eachmicrophone could be selected according to any well known gradient searchalgorithm.

FIG. 9A illustrates a microphone boom 320 having a first microphone 300and a second microphone 302 arranged therein. The first microphone 300includes a pressure sensitive surface 301 and the second microphone 302includes a second pressure sensitive surface 303. As shown in FIG. 9A,the first and second microphones 300 and 302 are arranged such that therespective pressure sensitive surfaces 301 and 303 are substantially180° apart from each other. The microphones 300 and 302 are furtherarranged so as to have a structural baffle 322 between the microphones.Such structural baffle 322 may be comprised of a structural memberadapted to provide an acoustical separation between the microphones.Alternatively, an acoustical baffling arrangement could be utilized inplace of a structural member. Furthermore, as shown in FIG. 9A, thefirst and second microphones 300 and 302, and in particular theirrespective pressure sensitive surfaces 301 and 303, are located within adistance or dimension b.

The first microphone 300 is adapted to receive acoustical sound such asspeech from a user and to convert such received acoustical speech into asignal corresponding to such speech. Such first microphone 300 may alsoreceive background noise which may exist. As is to be appreciated, suchbackground noise is combined with the speech from the operator and, assuch, the signal provided by the first microphone corresponds to boththe speech from the user and the background noise. On the other hand,the second microphone 302 is arranged within the microphone boom 320 soas to receive primarily only the background noise. More specifically,the pressure sensitive surface 303 of the second microphone 302 ispreferably arranged at an angle of substantially 180° from the pressuresensitive surface 301 of the first microphone 300. Further, aspreviously mentioned, the first and second microphones 300 and 302 havea baffle 322 arranged therebetween. Such baffle is adapted to minimizeor prevent any speech from the user from being received by the secondmicrophone 302. Furthermore, the first and second microphones 300 and302 are preferably arranged within relatively close proximity to eachother, that is, within the distance b. As an example, such distance bmay lie within a range of 0.10 to 0.50, preferably about 0.25 of aninch, or less. Suitable distance b may be determined by the skilledartisan from this disclosure, without undue experimentation and, theinvention is not necessarily limited to a particular value for b

FIG. 9B illustrates a microphone boom 330 having first and secondmicrophones 300 and 302 arranged somewhat differently than in themicrophone boom 320 of FIG. 9A. That is, as shown in FIG. 9B, the firstand second microphones 300 and 302 are located staggered side by siderelationship to one another. Further, a baffle 332 is provided betweenthe first and second microphones 300 and 302 so as to provide acousticseparation of the speech in a manner similar to that provided by thebaffle 322 of FIG. 9A.

FIG. 9C illustrates a boom headset assembly 400 incorporating the activenoise reduction device. As shown therein, such headset assembly 400generally includes a headband 401, a left case 402 having a left cover403 and a left cushion 409, a right case 404 having a right cover 405and a right cushion 410, a microphone boom assembly 413, and amicrophone boom 440. Such microphone boom 440 includes first and secondmicrophones 300 and 302 which may be arranged in a manner as previouslydescribed with reference to FIGS. 9A and 9B. Further, such microphoneboom assembly 440 includes an upper microphone case 406, a lowermicrophone 407, and the first and second microphones 300 and 302, and awindsock 408. FIG. 9C shows the arrangement of the active noisereduction apparatus comprising a sensor microphone 450 arranged relativeto the output transducer 460 and the acoustic filter 470 which can be afoam, pad, or the like. The preferred acoustic filter 470 is theSlo-flow foam, which is used to create an acoustical waveguide betweenthe speaker and the ear of the user as will be detailed in FIGS. 23A and23B. The function of the acoustic filter 470, which partially covers thesensor microphone 450 is to isolate the pickup microphone from theoutput transducer. The pickup microphone 450 does not have to be in thesame plane as the output transducer 460, but can be located above,below, tangential or adjacent to the plane of the output transducer 460.The portion of the acoustic filter 470 not adjacent to the sensormicrophone 450 also acts as an acoustical waveguide 475 located betweenthe output transducer 460 and the ear of the user. The acousticalwaveguide 475 is where the quiet zone is created as shown in FIG. 24.The acoustic waveguide 475 couples the user ear to the output transducerfor increased speaker efficiency. This positioning of the sensormicrophone 450 in a plane outside the plane of the speaker allows for aclose distance to the quiet zone for accurate phase agreement betweennoise and anti noise signals. In addition, the placement and orientationof the sensor microphone 450 minimizes microphone sensitivity lobepatterns in the direction of the speaker sound field. The sensormicrophone 450 used in the active noise reduction apparatus is anomnidirectional microphone, which is receptional to noise from allangles. This characteristics of the sensor microphone allows flexibilityin the positioning of the microphone with the earpiece of the headset.

FIG. 9D illustrates a side view of the boom headset assembly 400. Asshown therein, the left case 402 includes a circuit card assembly 412,which may contain circuitry utilized in processing the acoustic signalsas hereinafter more fully described, and further includes a cableassembly 411 for supplying signals to and from outside or host equipment(not shown). FIG. 9E illustrates a side view of the right case 404.

FIG. 9F shows an alternative embodiment of the sensor microphone 450location in the noise reduction apparatus. In FIG. 9F the sensormicrophone is located adjacent the output transducer 460 but notnecessarily in the same plane as the transducer. The acousticalwaveguide 475 covers a substantial portion of the speaker, with themicrophone 450 outside the quite zone area created within the waveguide475.

As previously described, the first and second microphones 300 and 302are preferably arranged within a distance b and are further arrangedsuch that the first microphone 300 receives both speech and backgroundnoise while the second microphone receives primarily just the backgroundnoise. Such background noise may originate as a pressure sound sourcefrom a location 304 as illustrated in FIGS. 7A and 7B. That is, as showntherein, such location 304 may be located at a distance r from a centerlocation between the first and second microphones 300 and 302 so as toform an angle θ therebetween. As a result, the distance between thefirst microphone 300 and the location 304 is approximately equal to thevalue [r-(b/2)(sin θ)], and the distance between the location 304 andthe second microphone is approximately equal to the value [r+(b/2)(sinθ)].

FIG. 8 illustrates a differential amplifier 500 which is adapted toprocess the signals produced by the microphones 300 and 302. As showntherein, such differential amplifier 500 includes an amplifier 310, anamplifier 312 and a summing circuit 314. The signal produced by thefirst microphone 300 is supplied to the amplifier 310 which is adaptedto provide essentially a unity gain to such signal and provide the sameas an output signal. Such output signal is supplied to one input of thesumming circuit 314. The signal produced by the second microphone 302 issupplied to the amplifier 312 which is adapted to essentially invert thereceived signal and to supply the same to a second input of the summingcircuit 314. The summing circuit 314 is adapted to add the receivedsignals together so as to produce a summed output signal e.sub.(out). Asis to be appreciated, such summed output signal e.sub.(out) represents asignal corresponding to substantially only the speech from the user.

FIGS. 10A and 10B illustrate the differential amplifier 500 of FIG. 8 inmore detail. That is, FIG. 10A illustrates a first arrangement of suchdifferential amplifier 500, and FIG. 10B illustrates a secondarrangement of such differential amplifier. Each of these schematicdiagrams will now be more fully described.

As shown in FIG. 10A, the signal produced by the first microphone 300 issupplied to an input terminal 600 and is supplied therefrom through aCapacitor C1 and a resistor R to an inverting input of an operationalamplifier (op-amp) V1A. The signal produced by the second microphone 302is supplied to an input terminal 602. Such input terminal 602 is coupledto a potentiometer RA which, in turn, is connected to ground. The inputterminal 602 is further coupled through a capacitor C2 and resistors R1and 2R to a non-inverting input of the op-amp V1A. Such op-amp isadapted to operate in a differential mode and provides an output signaltherefrom to a terminal 606 which, in turn, is coupled to the invertinginput of the op-amp V1A. Such output from the op-amp V1A is furthersupplied to a potentiometer 608 which, in turn, has one end connected toground. Such potentiometer 608 is coupled through a coupling capacitorC3 to a non-inverting input of an op-amp V1B. The output of such op-ampV1B is supplied to the base of a transistor 610. The emitter of suchtransistor 610 is coupled to a terminal 612 which, in turn, is coupledthrough a capacitor C4 to an output terminal 614. The summed signale.sub.(out) is supplied from the terminal 614. The collector of thetransistor 610 is coupled to a terminal 616 which, in turn, is connectedto a power supply (not shown) which supplies a voltage V+ to thecircuit. The terminal 616 is connected to resistors R3 and R4 which areadapted to provide a DC bias. The elements not specifically describedare connected as illustrated in FIG. 10A.

By utilizing the above-described circuit illustrated in FIG. 10A, theimpedance shown to the first and second microphones 300 and 302 issymmetrically balanced so as to minimize differential phase shiftsbetween frequencies. Further, the output signal from such circuit has arelatively low impedance.

FIG. 10B illustrates a second or alternate circuit of the differentialamplifier 500 of FIG. 8 as previously described. The circuit of FIG. 10Bis adapted to receive power through a source resistance from a powersupply (not shown). That is, the power for driving the circuit of FIG.10B is supplied from a power supply having a finite output resistance Runlike that supplied from a power supply having a zero output resistance(such as that of FIG. 10A). Otherwise, as is to be appreciated, thecircuit illustrated in FIG. 10B is substantially identical to that ofFIG. 10A and as such, will not be further described herein.

FIG. 11 illustrates a phase reversing circuit which may be utilized inplace of the circuits illustrated in FIG. 10A or FIG. 10B. Asillustrated therein, such circuit 700 generally includes the first andsecond microphones 300 and 302, the magnitude adjustment potentiometerRA, resistors R1 and R3, and capacitors C which are coupled asillustrated in FIG. 11. Each of the first and second microphones 300 and302 may include a field effect transistor (FET) in which the drain ofsuch FET may be considered a positive (+) and the source of such FET maybe considered a negative (-). The phase between such drain and source isapproximately 180°. For example, the drain thereof may have a phase of180°, whereas the source has a phase of 0°. As a result, each of thefirst and second microphones 300 and 302 includes two terminals, thatis, a positive (+) terminal and a negative (-) terminal. In the circuitof FIG. 11, the positive (+) terminals of the first and secondmicrophones may be the upper terminals thereof, whereas the negative (-)terminals of such microphones may be the lower terminals thereof.Further, the magnitude adjustment potentiometer RA may be adjusted orset during the initial assembly thereof or alternatively may be adaptedso as to be adjustable by an operator of the boom headset assembly 400of FIG. 9C. The output signal e.sub.(out) has a value of zero (0) whenan acoustical sound having the same pressure is received by both thefirst and second microphones 300 and 302.

FIG. 12 illustrates a circuit 800 which may be utilized in place of thephase reversing circuit 700 of FIG. 11. In the circuit 800, themicrophones 300 and 302 are oppositely charged. As a result, when theoutputs thereof are summed together, as when the first and secondmicrophones receive an acoustical sound having the same pressure, theoutput signal e.sub.(out) has a value of substantially zero (0). Theremaining portions of the circuit 800 are substantially similar to thoseof the circuit 700 of FIG. 11 and, as such, will not be furtherdescribed herein.

Thus, any of the circuits illustrated in FIG. 10A, 10B, 11 or 12 may beutilized in the present invention. Such circuits enable calibrationprocessing to be performed on the first and second microphones 300 and302 which may be electret-type microphones. Further, such circuits maybe included on a printed circuit (pc) board which may be installedwithin the headset assembly 400 as, for example, as on the pc board 412illustrated in FIG. 9D. Alternatively, such pc board may be included inother locations of the headset assembly 400 or alternatively may belocated on host equipment which is removed from the headset assembly400.

Thus, the present invention provides an assembly and, in particular, aboom headset assembly, which is adapted to reduce or eliminatebackground noise. The inventive apparatus can utilize first and secondmicrophones which act as a dipole, which may be matched by themanufacturer or by testing after manufacture, and which have a frequencyresponse which is essentially flat over the anticipated operating range.Further, such first and second microphones are preferably arranged suchthat their respective pressure sensitive surfaces are arranged at 180mechanical degrees from each other and are located in close proximitythereto as previously described. By so arranging the first and secondmicrophones, a sound (in particular a background noise) originating froma source which is located at a distance substantially greater than thedistance between the microphones, enables the sound from such soundsource to be received by both microphones simultaneously. As a result,no substantial phase differential therebetween occurs. Furthermore, thepresent invention enables the amount of noise cancellation to beadjusted either during the fabrication of the present boom headsetassembly or alternatively by an operator utilizing such assembly.

The boom microphone, for instance, of FIGS. 7A to 13B, can be based uponthe principles governing the directivity patterns of omni-directionalmicrophones in the near and far fields and the correct placement of themicrophone's pressure sensitive surfaces. The physical design of themicrophone as seen in FIGS. 9A and 9B is the determining factor in theS/N increase. Examination of these drawings shows that the microphonepressure sensitive surfaces are preferably placed at 180 mechanicaldegrees from each other, and provide the optimum separation of thesignal going to the voice microphone and noise microphone in the nearfield. This separation is a primary component in the determination ofthe signal in the S/N ratio. Basically, to achieve signals in the farfield is to add vectorially, at a desired point, the sound pressuresarriving at that point from all simple sources. A basic element of thisanalysis will be what is herein called the Doublet Sound Receiver.

The geometric situation is shown in FIGS. 7A and 7B. It is assumed thatthe distance r from the two receiving microphones (300,302) to point Aat which the pressure P originates is large compared with the separationb between the two microphones (300,302). The spherical sound wave frompoint A arriving at the receiving doublet will have traveled at adistance (r-b/2 sin θ) for microphone 300 and at a distance equal to thevalue (r+b/2 sin θ) for microphone 302. If r>>b as shown in FIGS. 7A and7B, the distance traveled by the spherical wave is r, and the output ofeach receiving microphone (300,302) is equal. If the summed outputs ofthe two microphones (300,302) are zero as in FIG. 8, then the associatedscale factors are equal. If their associated scale factors are notequal, any phase and amplitude of pressure can be the e.sub.(out) asshown in FIG. 8. Amplitude adjustment can be obtained electrically andis performed, but phase adjustment is not necessarily possible at allfrequencies. The requirement for phase adjustment is not requiredbecause reproducibility is inherent in the manufacture of themicrophones and they provide outputs of tracking phase with frequency.This method is described as a magnitude and phase microphone lobeconstructions which is the basis of the directional microphone of thepresent invention. These microphones will be capable of accepting soundsin certain directions better than in other directions.

The microphones just described are the dual of a doublet sound sourceand are similar to the theory of dipoles. In addition, if the spacing b,between the microphones is small (b<<X) compared with the wavelengths(λ) at any distance r, the two microphones essentially coalesce and theoutput at any angle φ will be zero for matched scale factors(magnitude/phase) at any frequency. In addition, the output at any angleθ can be electrically scaled and phased for the desired lobe intensityoutput. If b is not much smaller than r, the phase relationship betweenthe two microphones to an incoming sound wave is: ##EQU2## Whereb=spacing between sensor mic and noise mic

f=frequency in hz

v=speed of sound in/sec

φ=phase shift at a specific frequency

As can be seen from equation (1), this phase relationship is thetheoretical limit for the crossing of the near and far fields of thenoise canceling microphone. As the frequency changes at a fixed b, thephase changes, i.e.: at φ=90°, there may be no cancellation at all. Thisphase change, in the absence of acoustic filters can be a governingfactor in the bandwidth of the cancellation.

The embodiment of the invention of FIGS. 7A to 13B can be used on thefar field pattern of the microphones for noise cancellation. Thereduction of the effect of b, is performed by the use of an acousticdesign that tends to minimize or reduce to zero, dimension b, in FIG.9A, and that is modified to reduce the thickness of the probe as in,FIG. 9B. Both designs use the relationship between b and r (i.e.: b<<r).

To insure that the near field response is the desired one, electricalcircuit filters, such as the inclusion of a low pass filter 916 as shownin FIG. 14, allows only voice band frequencies ^(e).sub.(out) =(mic₁-mic₂)^(k) /_(s+w) to be present in the output and keep the restrictionson b and λ within practical constraints.

If the number of elements is increased, and an array of microphones isimplemented as shown in FIG. 15, the lobe patterns forcancellation/reinforcement is the side lobes (θ=90°, θ=270°) and thelobe patterns at θ=0° and θ=180° are increased, the greater the numberof elements in the linear array, the sharper the θ=0°+θ=180° lobepatterns. FIG. 15 shows the microphones 922 uniformly spaced along thex-axis to facilitate analysis but spacing need not be uniform. Theintermediate pair of elements M₂ and M₃ are spaced apart a distance b,and the outer pair of elements M₁ and M_(N) are spaced apart a distanced. A sinusoidal wave is created when sound pressure is applied to themicrophones and is incident of the array of elements or microphones. Thedirection of propagation of the wave creates an angle θ with the x-axisintersecting the same at point 918, midway between elements M₂ M₃. Theamplitude of the wave (not shown) is a measure of the instantaneoussound amplitude 304 at that point. The distance d is the linear distancebetween a first microphone and nth microphone. These statements arebased on similar microphones, and the use of appropriate electricalfilters such that when r is not much greater than b, the electricalfilter allows the microphone dipole to be mathematically manipulated forlobe construction. The concept of lobe construction is known to oneskilled in the art.

After the array has been utilized to adjust the receiving lobes to θ=0°and θ=180°, the 180° lobe will be greatly attenuated by placing thelinear array into a cylinder as shown in FIG. 16A of radius ^(a) /₂,where the value of a is the sound input with the 180° side sealedphysically with an inside sound absorbing pad 905 to prevent theproduction of any standing waves. Slots or apertures will be added in aplane parallel to the microphone array principal axis to insure thedipole action to oncoming acoustical signals is at θ=90° and θ=270°.

It should also be noted that the end microphones of the array, namely1+n of FIG. 16A are displaced by a linear distance d along the axis ofthe cylindrical baffle 905, which acts as an acoustic resistance. Sincethis distance is large enough, the microphones are independent andcauses a further enhancement of the sound along the θ=0° axis, whichincreases directionality.

FIG. 16B is similar to FIG. 8 in that the differential amplifier isadapted to process a linear array of signals produced from microphones900 to the nth microphone to an amplifier 910, with the output inputtedto an summing circuit 914, where the output signal is transmitted to afinal summing amplifier used in such circuits as described in FIGS. 7Ato 13B. The output signal corresponds to substantially only speech fromthe user.

In FIGS. 10A and 10B, the circuit is similar to the circuits utilized inthe telephone embodiments above-described with respect to FIGS. 1 to 5.In this circuit U1A is utilized for the subtraction, and U1B is utilizedfor output interfacing.

The phase reversing circuit is shown in FIG. 11. This circuit willprovide two signals at points A and B 180 degrees out of phase withidentical sound signals in microphones 1 and 2, if the microphones arematched parts (by manufacture). This output can be adjusted foramplitude matching at a reference frequency by adjustment of RA inconjunction with capacitor C. The signal at e_(out) is the noisecanceled output when the microphones are placed in the appropriatemechanical environment mentioned previously.

Analysis of the circuit of FIG. 11 can be shown to provide the followinginformation. The output at A is at the Source of the internal FETcontained within the microphone (preferably electret) such that itsoutput is at an electrical angle of 0 degrees with the input pressuresignal, while the output at B is from the Drain of the internal FETcontained within electret microphone and its output is at an electricalangle of 180 degrees with the input pressure signal. When these two farfield signals are summed together in a voltage mode, the output is zeroif the amplitude is adjusted by potentiometer RA at a referencefrequency and the magnitude response is flat across the frequencyspectrum.

In the circuit in FIG. 12, the oppositely charged microphones providetwo signals at A and B 180 degrees out of phase with identical soundsignals at microphones 1 and 2. This phase reversal is accomplished byvirtue of opposite charging during manufacture of the electretmicrophone condenser plates.

All of the other characteristics are as previously stated for the phasereversing circuit. Circuits of the type found in FIGS. 11 and 12 providefor electrical subtraction without the need for using an op amp.

In addition, the boom microphone/system of the invention is optimallydefined by the location of the microphone's pressure surfaces,preferably 180 degrees in the case of the boom microphone, butcancellation will occur because of the subtraction type system definedin FIGS. 10A-12 at all angles. In fact, when the microphone pressuresurfaces are at 0 with respect to each other, total cancellation couldbe theoretically obtained but no audio signal would be transmitted. Thesystem of the present invention can rely on the directivity patterns ofthe microphones in the near and far fields, orientation of theirpressure sensitive surfaces, and the electrical process of subtraction.

The typical circuits that can be utilized for subtraction are shown inFIGS. 10A-12.

The boom microphone headset device of the invention (e.g. FIGS. 7A-13B)can provide for computer voice recognition. The boom microphone headsetprovides superior rejection of unwanted background noise and excellentvoice response. The boom microphone headset can be configured to becompatible with all Sound Blaster™ audio cards. All other audio cardinterfaces are also easily accommodated.

The inventive boom microphone headset (e.g., FIGS. 7A-13B) coupled withthe latest in high quality voice recognition software advances computercontrol with Voice to a reliability and user friendliness level equal tothe keyboard and mouse. With the present invention, voice recognition isno longer confined to quiet closed door offices, but can be used inreal-world noisy environments such as hotel lobbies, hospital emergencyrooms, manufacturing facilities and noisy office areas. Thus, the boommicrophone headset can interface with computers, telephones or otherequipment in the real world or, the boom microphone (without headset)can be employed in various voice recognition applications.

The inventive boom microphone headset is designed to be sensitive todistance from the sound source. Arbitrary sound fields which emanatefrom more than a few inches away from the microphone are substantiallycanceled by up to 30 dB (3200%).

The inventive boom microphone headset preferably is connected to 3 metercable which terminates in a 3.5 mm miniature plug (not shown). Toconnect it to the sound card, the user simply inserts the miniature pluginto the Microphone input jack of the sound card (not shown). Theinventive boom microphone headset then is placed on the user. Theheadset preferably has two features to help position the microphone inthe proper position for reliable voice recognition: (1) adjustabletemple pads on both the right and left sides and (2) an adjustable flexboom. The microphone at the end of the flexible boom preferably has awhite or other color-coded dot indicating the voice-side of themicrophone which should be adjusted to directly face the mouth. Properclose talking into the invention is helpful for accurate operation.Preferably the distance from the mouth to the microphone should be inthe range of about 1/4 to about 3/4, preferably about 1/2 inch or less.

As to use with the Sound Blaster™, it is important to disable AutomaticGain Control (AGC) on the sound card before using the present inventionin noise canceling applications. If AGC is ON, it will defeat the noisecancellation properties of the microphone by automatically turning upthe input audio volume when the user is not speaking. The AGC can bedisabled on Sound Blaster™ cards by running Creative Mixer™ and clickingon the AGC software control under "Recording Settings . . . ". The inputaudio gain is easily tailored to the target application using theCreative Mixer™ program. Testing of the inventive microphone headset caneasily be performed by using the record and playback features of theCreative Labs Wave Studio™ program.

FIGS. 13A and 13B are active noise cancellation curves of the boommicrophone in a headset embodiment with FIG. 13A, top line, representingnear field response and FIG. 13A, bottom line, representing far fieldresponse. In FIG. 13B, the top line represents the close talkingresponse and the bottom line represents the background noise response.Typical specifications for an embodiment of the inventive boommicrophone headset include

Frequency Resp: 20 Hz to 10 kHz

Output Impedance: Low Impedance (Capable of driving 560 ohm)

Sensitivity: -47 dB±2 dB (0 dB=1 v/Pa@ 1 kHz, 5V)

Operating Voltage: 2V to 10V

Current: <1 mA (power supply 5V)

Electrical S/N: 60 dB (minimum)

Noise Cancellation: See FIG. 13B

Cable Type: Non-detachable, shielded

Length of Cable: 3000±50 mm

Plug Type: 3.5 mm stereo miniature. male

Weight: 56 gm (without cable)

Using interpretation of speech intelligibility AI and ANSI S3.5-1969, aboom microphone headset of the invention and a standard (prior art)dynamic noise canceling microphone were tested and the results were asfollows:

Articulation Index: Invention Boom Microphone

    ______________________________________                                                              Weight                                                  1/3 Octave Band                                                                         S/N (dB)    Factor (BW                                                                              Articulation (1)                              Center Freq. (Hz)                                                                       [NPR-FPR]   Corrected)                                                                              Weight (W)                                    ______________________________________                                         200      26.6        0.00046   0.01219                                        250      24          0.0012    0.0288                                         315      22          0.0012    0.0264                                         400      20.5        0.0016    0.0328                                         500      17.5        0.0016    0.028                                          630      16          0.0023    0.0368                                         800      15          0.0023    0.0345                                        1000      12          0.0028    0.0336                                        1250      15          0.0035    0.0525                                        1600      9.5         0.0043    0.04085                                       2000      9           0.0044    0.0396                                        2500      5           0.0039    0.0195                                        3150      1           0.0039    0.0039                                        ______________________________________                                    

Articulation Index: Standard (Prior Art) Dynamic Noise CancellationMicrophone

    ______________________________________                                                              Weight                                                  1/3 Octave Band                                                                         S/N (dB)    Factor (BW                                                                              Articulation (1)                              Center Freq. (Hz)                                                                       [NPR-FPR]   Corrected)                                                                              Weight (W)                                    ______________________________________                                         200      25.5        0.00046   0.01173                                        250      18          0.0012    0.0216                                         315      12          0.0012    0.0144                                         400      11          0.0016    0.0176                                         500      5.5         0.0016    0.0088                                         630      3           0.0023    0.0069                                         800      0           0.0023    0                                             1000      4           0.0028    0.0112                                        1250      3.5         0.0035.   0.01225                                       1600      5           0.0043    0.0215                                        2000      2.2         0.0044    0.00968                                       2500      3           0.0039    0.0117                                        3150      2           0.0039    0.0078                                        ______________________________________                                    

Interpretation of speech intelligibility using AI and ANSI S3.5-1969shows an accuracy level of 93% for the present invention versus anaccuracy level of only 45% for the Standard Dynamic Noise CancelingMicrophone. A comparison of this data reflects a reduction in errorratio of approximately 8:1 by the present invention (i.e., AI 45% stdDynamic Mic, Noise Canceling AI 93% by present invention). Furthermore,additional AI is expected when constants are corrected to be active downto 50 cycles and below. Literal evaluation of the AI calculation statesthat for every 100 words spoken, the present invention will commit 7errors, and Standard Dynamic Microphones will commit 55 errors. All dataand calculations were collected and performed at Andrea ElectronicsCorporation. Both microphone systems were tested at Andrea ElectronicsCorporation under the same conditions.

FIG. 17A shows a block diagram of a prior art amplifier plus amicrophone 1004 having three terminal circuit configuration foroutputting signals or power supply inputs from standard on-line powerused in any type of microphone device, which includes an amplifier 1004connected in telephone headset or the like. The three terminalconfiguration comprising a power input terminal 1001, a voltage outputterminal 1005, and a ground terminal 1003. In an effort to reduce thesimplicity of a signal processing circuit, the block diagram of FIG. 17Bshows a microphone plus amplifier configuration of the present inventionhaving a two terminal microphone. Terminal 1005 is adapted to receivethe power input 1001, which is a DC signal received from a power supply,thru a resistor 1006 and transmits the audio output, an AC signal, onthe same terminal, namely 1005, while terminal 1003 is grounded. Thus,at terminal 1005 the DC power is supplied for the active noisecancellation system described herein, and the audio signal generated bythe microphone 1004 are concurrent. This point 1015 is the collector oftransistor 1050 shown in FIG. 18.

FIG. 18 illustrates the basic circuit for processing signals obtainedfrom the first and second microphones 12 and 14, respectively, so as toprovide an audio signal output 1015 generated by the microphone signals1060 outputted from the operational amplifier 1070, and a DC powerinputted at terminal 1020 thru a resistor 1005, which signal and powerbeing concurrent on the same terminal. That is, the audio signal isoutputted and DC power is being supplied on the same terminal 1015. Amethod to separate the DC and AC at point 1035 is desired, which ispreferably the resistor 1030 coupled with the capacitor 1040. The DCobtained at point 1035 is used to power the operational amplifier 1070as well as the microphones 1060, and the terminal at point 1015 is usedto output the audio signal. At point 1015, the audio signal istransmitted from the collector of a transistor 1050, and the AC signalis separated from the DC signal by resistor (R) 1030 coupled with thecapacitor (C) 1040. This resistor coupled with the capacitor creates alow pass filter that provides the AC (audio) and DC (Power Separation),which filters the AC from the DC. The AC and DC signals at point 1015are separated at point 1035 where only the DC is allowed to existsbecause of the filtering action of resistor 1030 plus capacitor 1040arrangement. The audio output at point 1015 is generated by the ACsignals of the microphones 1060. This separation is the preferred meansfor operation of the present invention due to the microphone subtractionthat is required. At 1015, the AC signal from the input and the DC powerare transmitted to the resistor 1030. The AC signal is then filtered outby the resistor 1030 coupled with the capacitor 1040, and the DC is usedto power the circuit 1070. The audio signal, which is an AC signal istransmitted as an output signal at 1015.

FIG. 19 illustrates a different embodiment of the two wire microphoneswhere the power supply 2020 is inputted through resistor 2005 and the DCsignal generated is transmitted on the same wire lead 2015. Therefore,the audio AC signal outputted by the collector of the transistor 2050 issuperimposed on the dc signal at the same terminal at point 2015, thecombination signal being supplied to resistor 2030. The AC component isthen filtered out by the resistor 2030 coupled with the capacitorcircuit 2040, with the remaining DC signal used to power the circuitrydesignated by 2055 including the operational amplifier 2070. The audiooutput signal transmitted at 2015 is generated by AC signals outputtedby the two microphones 2060, and the signal is processed through thecircuitry 2055 and the operational amplifier 2070, which understandingis apparent to those skilled in the art, and outputted at the sameterminal at point 2015. This two-wire microphone circuit is more simplerthan the prior three wire microphones circuitry, involving an inputterminal, voltage output terminal, and ground terminal as shown in FIG.17A. This two-wire microphone circuitry is utilized in any of thecircuitry described in FIGS. 4-5, 10-12, where the signal is generatedby at least two microphones. In addition, the two wire microphonecircuitry is also utilized in the far-field directional microphonesshown in FIGS. 14-16.

FIG. 20 is an alternative embodiment of FIG. 19 employing a two-wiremicrophone circuitry, having the audio signal output and power inputtransmitted at the same lead terminal 3015. The AC signal generated bythe collector of the transistor 3050 is outputted at same lead terminal3015 as the DC power signal, which is generated by the power supply 3020fed through the resistor 3005. Therefore, the AC signal is superimposedon the input DC signal at 3015, which combined (AC plus DC) signal istransmitted to resistor 3030. It is here that the AC component of thecombined AC plus DC signal is filtered out by capacitor circuit 3040coupled with the resistor 3030, with the DC signal remaining used todrive the remaining circuitry 3055 and the operational amplifier 3070,the understanding of this circuitry 3055 is apparent to those skilled inthe art. The AC audio signal outputted by the microphone circuitry 3055and 3070 is transmitted at the same terminal at point 3015 as the DCsupplied for the active noise cancellation system described herein.

The individual circuit component without reference designations in FIGS.19 and 20 are connected as shown and will not be discussed further,since the connections and values are apparent to those skilled in theart and not necessary for an understanding of the present invention.

FIG. 21 illustrates the closed loop system incorporated into the activenoise reduction of the prior art. The governing equation is: ##EQU3##whereby P represents the audits sound pressure, S is the audio signal,H₁ is the high pass filter, H₂ is the speaker at the headset, B is thevariable gain/phase control and N is the noise at the pickup microphone.The N noise component 1210 is zero at the P, pressure output 1200because of the very large transmission gain (1+BH₁ H₂). In order thatthe desired S audio signal 1220 which is inputted as an electricalsignal is not canceled, two summing nodes (1230, 1240) are added. Theaudio signal 1220 is inputted into two summing nodes (1230, 1240) tocreate the signal transmitted to the user ear. The first summing node1230 adds the negative feedback signal to the desired input audiosignal. However, the signal feedback from the microphone contains thedesired audio signal as well as the ambient noise signal, the noisesignal being the desired signal to be canceled. The feedback signal issubtracted from the desired input audio signal 1220 to create theanti-noise signal 1250, with no audio signal contained. A second summingnode 1240 adds the audio signal 1220 into the loop to transmit to thespeaker.

FIG. 22 relates to the active noise reduction system of the presentinvention, which is governed by the following equation P=AH₂ -(N+D)BH₁H₂ +N₁. In the equation, P represents the sound pressure 1330, Arepresents the standard audio transmission 1340; H₂ represents thespeaker at the headset 1370; N is the noise at the pickup microphone1315; D represents the very low frequency disturbance 1320; B is thevariable gain/phase controller or calibration pot 1350; H₁ representsthe high pass filter 1380 and N₁ is the noise 1390 at the quiet zone atthe ear of the user. The active noise reduction system is comprised ofan open loop circuit having the following components: an audio signal1340; a sensor microphone 1310 able to detect and cancel noise 1315; aoutput transducer 1370 located near the user's ear; a variablegain/phase controller 1350 to adjust the amplitude of the anti-noise1315; a summing node 1360 to sum the anti-noise signal and audio signal1340; a high pass filter 1380 to prevent mechanical vibration inducedfrequency disturbance components 1320 from reaching the outputtransducer. The system detects ambient noise 1315 by the sensor orpickup microphone and applies electroacoustical processing to produce anacoustical signal for canceling out the ambient noise. This system maybe used to cancel all noise, so as to obtain a signal representingspeech, which is the desired signal to be heard through the ear of theuser.

The active noise reduction system in FIG. 22 cancels noise at a specificpoint in space by sensing that noise 1315 with a sensor microphone 1310and producing an anti-noise signal which is of the same magnitude but180° out of phase with the input noise N₁ signal 1315. By adjusting thevariable gain/phase controller 1350 to create an anti-noise signal ofthe same magnitude, but 180 degrees out of phase with the noise signal1315, and summing the inputted audio signal 1300 and anti-noise signalat one summing node 1360 to yield an anti-noise plus audio signal, theambient noise is attenuated without the input audio 1300 being degradedby mechanical or vibration induced microphone signals.

The variable gain/phase controller or calibration pot 1350 of thepresent invention remotely tunes or balances the sensor microphone tothe output speaker in the headset or any other communication apparatushaving an earphone with a microphone means to transmit intelligiblespeech. The controller or calibration pot balances the gain and phase ofthe frequency response of the noise signal detected from the pickupmicrophone 1310 to match the gain and phase of the noise componentsignal at a predetermined point in space. This point in space is inproximity to output transducer located in the earphone of the headset.Often the phase component of the noise signal is relatively flat due tothe close proximity of the microphone to the output transducer, and thefrequency response is linear to the output transducer. By independentlycalibrating the gain and phase microphone and signals in outputtransducer of the headset in an active noise reduction apparatus, tomatch the gain and phase in the earphone, a theoretical zero is yieldedat a point in space, or a quiet zone. Basically, the ambient noisesignal is inverted to an anti-noise signal by the adjusting the gain andphase to be canceled with a noise component located at a predeterminedpoint in space, i.e. the quiet zone. The gain phase controller orcalibration pot of the present invention provides for flexibility in theutilization of headsets with any communication system, i.e. flightcontrols system, computer interface, telephone network, or the like, asthe headsets are readily adaptable for use in any communication system.

Therefore, the desired audio input signal 1340 is transmitted to theoutput transducer in the earphone without the disturbance of backgroundnoise. The output audio signal that reaches the ear of the user is ofthe formula P=AH₂, which is the desired audio transmission that wasoriginally inputted to the headset or the like. The reduced complexityof the open loop design allows for all types of noise to be canceled,namely syndromes, repetitive, and transient.

The noise cancellation system 1300 described in FIG. 22 can becharacterized as an open loop system without closed loop electricalfeedback compensation. The system in FIG. 22 is capable of driving aspeaker by employing a method of adjusting the parameters of the openloop system by the variable gain/phase controller 1350 and/or correctacoustic filtering of the sensing microphone. The preferably acousticfiltering is utilizing specifically selected foam, the most preferableis the Slo-Flo foam as shown in FIG. 23B. The positioning of the pick-upmicrophone 1310 relative to the speaker and the foam creates anacoustical waveguide 1400 as shown in FIG. 23A between the headsetspeaker 1370 or any other suitable speaker and the ear of the user. Thismicrophone positioning channels the desired audio signal to the user earcanal and isolates any audio signal generated from the noise detected bythe pickup microphone 1310. The pickup microphone 1310 is preferablyplaced outside the plane of the speaker and outside the acousticwaveguide, but close in distance to the ear canal of the user where thequiet-zone field as described in FIG. 24 is created. The quiet zone inFIG. 24 maintains phase agreement between the noise signal picked up bythe microphone and the anti-noise signal located in the acousticalwaveguide, located behind the speaker. The present invention is notconcerned with the microphone and speaker being in substantially thesame plane. They may be in the same plane; but, they do not need to bein the same plane. Rather, the noise reducing device of the presentinvention is concerned with creating an acoustical waveguide by use ofacoustic filters between the device speaker and the ear of the user, forchannelling the audio input signal to the user's ear canal withoutdisturbance from any noise detected by the pickup microphone 1310, whichhas been canceled. No noise cancellation may occur when the phasedifference is 90 degrees, or 1/4 wavelength and reinforcement wouldoccur at 180 degrees or 1/2 wavelength. The equation that governs thisdistance is: ##EQU4## where .O slashed.=phase at specific frequency

l=distance

f=frequency

v=speed of sound (in/sec)

The H1 transfer function 1380 of the open loop equation as shown in FIG.22 is a high pass filter. Over the frequency range, the high pass filteris active at H1=1, and is inactive at H1≈0. The purpose of the high passfilter 1380 is to reject the low frequency mechanically inducedtransients in a regular manner so that the active noise reductionperformance is continued in a regular manner, but not used as a leadstabilization network as in the standard closed feedback systems. In theopen loop equation governing the present invention in FIG. 22 is theplacement of the breakpoint, which provides optimum performance of thedevice. The breakpoint is where the amplitude goes from zero (0) to theleading edge of the pass band. In the frequency range below thebreakpoint, H1≈0 and the product (D×H1)=0. The adjustment procedure isthen as follows:

P=AH2-(N+D)BH1H2+N1becomes

P=AH2-NBH1H2+N1 if

NBH1H2=N1 by B adjustment

P=AH2, which is desired Audio Signal.

Above the breakpoint, H1=1 and D=0. The product H1D=0 and by similarreasoning, P=A×H2, the desired audio signal. The correct placement andslope of the breakpoint transition provides for the most preferredactive noise reduction and optimum disturbance performance. Due to theinherent stability of an open loop system, the mechanical and vibrationlow frequency signals can be electrically filtered out without addingsignificant complexity to the circuitry and having to deal with closedloop stability requirements.

Measurements of the transfer function of FIG. 22, have shown that theopen loop concept is in embodiment of the present invention. Thetransfer function from N to P has shown a magnitude of less then 1, andif the system was closed loop with G=1, H=1 system the maximumcancellation is 50%. ##EQU5##

An open Loop system in fact would approach 100% cancellation under theseconditions.

There is no need to add a second summing node in the present inventionas shown in FIG. 22. However, the prior art closed loop noisecancellation systems double adds the audio to reduce the effect ofpick-up microphone subtraction as detailed in FIG. 21. The presentsystem shows no audio reduction with only a single audio summation node1360 in FIG. 22, as the present invention is concerned with cancelingonly noise detected by the microphone 1310 and the microphone isindependent from the audio signal being transmitted to the speaker forobtaining the desired audio signal 1340.

A mechanical diagram of the active type noise cancellation device of thepresent invention is shown in FIGS. 23A and 23B. The acoustic waveguide1400 shown in FIG. 23B, may perform anyone of the functions of 1)channeling sound between speaker 1410 and user ear 1430 and 2) isolationof sound emanating from the speaker/quiet-zone 1440 at the user's ear1430 to the noise pickup microphone 1420 at the edge of the speaker.Elements of the mechanical design of the present device may be any of 1)close distance between the noise pickup microphone and anti-noisesignals, 2) placement, orientation and isolation of the microphone tominimize microphone sensitivity lobe patterns in the direction of thespeaker sound field, 3) use of an acoustic filter or baffle 1440, suchas the preferred foam, Slo-Flo, but other suitable material may be usedto create the acoustical waveguide. The acoustical waveguide isolatesthe noise pickup microphone from the speaker as shown in FIG. 23A tocreate the quiet-zone 1440 in front of the user's ear as shown in FIG.23A for increased speaker efficiency. The acoustical waveguide acts as areceiver for the anti-noise signal, generated from the signal processingmeans of the electrical signals from the microphone. This anti-noisesignal creates the quiet zone of the acoustical waveguide. The devicewith acoustic waveguide can be applied to headphones of all typesincluding open back, that is, no ear passive earcups, or closed backtype headsets, that is with passive noise attenuation earcups or anyother suitable headsets utilizing a receiver and transducer.

FIG. 24 is an evaluation of a quiet zone in space whereby the firstvector 1500 is the noise vector and the second vector is the anti-noisevector 1501 produced by the active noise reduction system of the presentinvention. The two vectors create an angle θ, which phase and magnitudeis attenuated by the variable gain/phase controller, thereby controllingand thereby reducing the anti-noise processing induced distortion of thedesired electrical input signal; which is converted to an acousticsignal and transmitted to the ear of the user.

FIG. 25 illustrates a schematic diagram, which embodies the active noisereduction system described in FIG. 22. The schematic diagram comprises apickup microphone 1600, a speaker 1650, a variable gain/phase controller1610, an signal convertor 1620, a summing amplifier 1630, a poweramplifier 1640, an anti-noise output signal 1660. A standard audiosignal 1605 is inputted to the user of the headset. The sensor pickupmicrophone 1600 detects ambient noise and creates an electrical signal.This signal is inputted into an electro-acoustical processing unit,which is comprised of the variables gain/phase controller 1610, thesignal convertor 1620 and the summary amplifier 1630 to produce anacoustical signal for canceling out the ambient noise, referred to as ananti-noise signal outputted at 1600. This anti-noise signal is placed infront of the speaker at some point in space to achieve a quiet zone asshown in FIG. 24. The desired input audio signal 1605 is isolated ordisposed from the microphone, hence, the input audio signal is notassociated with the ambient noise detected by the pickup microphone1600. The original input audio signal 1605 is able to be transmittedthrough the speaker into quiet zone, without noise disturbance.Therefore, in the open loop system of the present invention, the audioinput signal generated in the active noise reduction system is notdisturbed by the ambient noise, which noise is detected and reduced bythe sensor pickup microphone circuitry comprising: the variable gainadjustor 1610, the communication audio 1620, the summing amplifier 1630and the power amplifier 1640. Hence, the audio input signal 1605 doesnot have to be double added as required in a close loop system.

FIG. 26 illustrates a different embodiment of FIG. 25 which incorporatesan open loop system for active noise reduction by using a sensor pickupmicrophone 1700, a variable gain/phase controller or calibration pot1710, communication audio signal 1720, a summing amplifier 1730 and apower amplifier 1740, and speaker 1750. The operation of this circuit isdescribed in FIGS. 22 and 25.

FIG. 27 is a preferred embodiment of FIG. 25 with a high pass filtercircuit 1840 added to the active noise cancellation system to reject ona regular basis the low mechanically induced transients so not tointerfere with the active noise reduction performance. The high passfilter circuit 1840 is not to be used as a lead stabilization network asin the standard feedback system. FIG. 27 also contains a saturationreduction 1860 for clipping diodes. The function of the clipping diodesin FIG. 27 is to prevent the output transducer from reaching itsphysical limits. That is, it limits the amplitude of the signal inputtedto the transducer to prevent the speaker or output transducer fromexceeding its physical excursion.

The individual circuit component without reference designations in FIGS.25-27 are connected as shown and will not be discussed further, sincethe connections and values are apparent to those skilled in the art andnot necessary for an understanding of the present invention.

FIG. 28 is an enhancement of the active noise cancellation and noisereduction system utilized in a headset having a "talk-thru" capability.

The enhancement in FIG. 28 is achieved by including the followingfeatures: an automatic audio microphone transmission in the active noisecancellation system to sense speech ("VOX circuit") 1950, the ability totransmit a portion of the received microphone signal to earcup of thespeaker ("sidetone circuit") 1960, 1907, 1970 and 1930, the capabilityof converting an active noise cancellation microphone to a standardomni- directional microphone by removing the voice microphone from thecircuit, and increasing the gain of the noise microphone amplifiers("talk thru") 1930. With the "talk thru" feature, all audio (1990) fromthe external area is transmitted to the earcup speakers 1980 by anincreased gain sidetone channel 1907. The enhanced active noisecancellation and noise reduction headset in FIG. 28 comprises: activenoise cancellation microphone 1900 that detects only the audio signalsand active noise cancellation microphone 1901 that detects the audiosignal and background noise; a first S1A switch 1910 having a noisecanceling mode and talk-thru mode or position; a second S1B switch 1925having a noise canceling mode and a talk-thru mode; a third S1C switchhaving a noise canceling mode and a talk-thru mode; a Push to Talk (PTT)switch 1920 having a hot mike position and simulated speech signal mode;active noise cancellation microphone amplifier 1940; VOX circuit 1950,speech signal 1955; transmission gate 1945; buffer amplifier 1935; audiosystem 1915; scaling amplifier 1970 having a gain control function 1907,a sidetone signal 1960 inputted at 1906; earcup speaker 1980; externalaudio system 1990; and an active noise reduction system as preferablydescribed with respect to and shown in FIGS. 21-24. This headset in FIG.28 operates in either a noise canceling mode or a talk-thru mode.

In the noise canceling mode, the switch Si is in the "N" position 1910,and the active noise cancellation microphones 1900 and 1901 areoperating as previous described herein. The PTT switch (Push to Talk)1920 is not activated in the noise canceling mode. The VOX circuit 1950which is connected to the microphone output of the microphone amplifier1940 at A 1945 monitors the microphone output signal of the amplifier1940. The VOX comprises an attack time (turn "on time" averaging circuitof audio), and a release time (turn "off time" averaging circuit ofaudio) which is adjusted to minimize response to spurious signals and tokeep the microphone "on time" to a minimum in the noise canceling mode.This in reality will increase the system's speech to noise ratio "S/N".When the VOX 1950 has determined that the signal at the output of themicrophone is useable audio, it will activate the "speech signal" atpoint C 1955, which will enable the transmission gate 1945 and allow themicrophone audio outputted from the active noise cancellation microphoneamplifier means 1940 into the buffer amplifier 1935 and then to theaudio system at E 1915. In addition, this audio 1915 is sent to thescaling amplifier 1970 at W 1906 to provide a sidetone signal to theearcup speakers 1980 when the third switch S1C 1960 is at N, the noisecanceling mode. The scaling amplifier 1970 can also simultaneouslyaccept an input from an external audio system 1990, i.e. distinctivesounds from the surroundings, such as sirens, bystander's voices, orother external sounds not being transmitted by the microphones 1900 and1901. The composite signal at the earcup speaker 1980 is the linearaddition of sidetone 1960 and external audio 1990.

In the talk-thru mode, switch Si is placed in the talk-thru mode orposition shown as 1930 for the first, second and third switches (1910,1925, 1930). The voice microphone 1900 is disabled. The voice and noisemicrophone 1901 is enabled. The microphone amplifier 1940 output at A1905 is the noise omnidirectional microphone 1901 output. In thetalk-thru mode, the VOX circuit 1950 is bypassed by the second S1Bswitch 1925 placed in the talk thru position, which allows the directoutput of the noise omnidirectional microphone signal 1901 to the bufferamplifier 1935 at D 1903 and then outputted to the audio system 1915 atE. As a result, no speech signal is inputted to transmission gate 1945output at B 1904 because the gate 1945 is disabled. The gain controlfunction 1907 of the scaling amplifier 1970 is increased at W 1906, bythe action of switch S1C 1960, at the talk thru position 1930. Thus, thesidetone signal outputted from the active noise reduction system(described in FIGS. 21-27) is increased at the speaker 1980.

As a result of the arrangement described in FIG. 28, and without thephysical addition of any other microphone elements, a talk thru featurehas been added to the headset described with respect to and shown inFIG. 9-27. This allows audio transmission of voice to be heard in theearphone speaker 1980 without removal of the headset. This enhancementprovides the headset user the option to continuously wear the headsetsto receive the audio transmitted from the microphone signals, as well asany other distinctive external noise in the surrounding environment.This external noise can be any sounds, such as explosions, emergencysirens or bystanders speaking to the headset user. The enhancementavoids the awkwardness and inconvenience encountered in removing theheadset constantly throughout the day to hear external noises not beinginternally communicated thru signals in the headset. The user now isable to hear internal audio signals and external audio from thesurrounding environment while wearing the active noise reduction andnoise cancellation apparatus described with respect to and shown in FIG.28.

Further, although the above-described embodiments of the presentinvention have been described for use with telephone handsets and boommicrophones and the like, the present invention is not so limited andmay be used with numerous other devices such as intercom systems,telemetry, acoustic surveillance microphones, directional microphonesand so forth. Further, the invention can be utilized in voicerecognition and/or verification systems such as systems for access tophysical facilities, computer programs, computers or automatic tellermachines and the like. Additionally, the present invention may be usedwith processing devices operating in accordance with predeterminedprocessing algorithms, as described in U.S. Pat. No. 5,251,263, whichhas a common assignee with the present application, and which is herebyincorporated by reference; however, such is not believed necessary tothe invention.

Furthermore, although preferred embodiments of the present invention andmodifications thereof have been described in detail herein, it is to beunderstood that this invention is not limited to those preciseembodiments and modifications, and that other modifications andvariations may be affected by one skilled in the art without departingfrom the spirit and scope of the invention as defined by the appendedclaims.

What is claimed is:
 1. A transducer for use in a noise cancellationapparatus for reducing background noise comprising:a housing havingfirst microphone means for receiving a first acoustic sound composed ofspeech originating from an operator operating said apparatus andbackground noise, and for converting said first acoustic sound to afirst signal, and second microphone means arranged at a predeterminedangle .O slashed. in close proximity with respect to said firstmicrophone means for receiving a second acoustic sound composed ofsubstantially said background noise and for converting said secondacoustic sound to a second signal; said first and second microphones areconnected to a differential amplifier means of the noise cancellationapparatus so as to obtain a signal representing substantially speech;the amplifier means for receiving acoustic sounds from each microphoneand having a first terminal and a second terminal, wherein the secondterminal is grounded; a transistor means for receiving and amplifying anAC signal representative of the audio input from each microphone; andmeans for filtering the amplified AC signal from the DC signal, so thatthe DC signal powers the amplifier means.
 2. The transducer according toclaim 1, wherein the amplifier means is an operational amplifier.
 3. Thetransducer according to claim 1, wherein the means for filtering the ACsignal from the DC signal is a resistor coupled with a capacitorcircuit.
 4. An active noise reduction apparatus for reducing ambientnoise in the vicinity of an eardrum, comprising:a housing for receivingan input audio signal; an output transducer located in the housing; aninput transducer for detecting and reducing ambient noise, the inputtransducer not located in substantially the same plane as the outputtransducer; signal processing means to reduce the ambient noise detectedby the input transducer; an acoustic means including an acousticwaveguide for transmitting the input audio signal to the eardrum withoutdisturbance of the ambient noise and having low pass filtercharacteristics with a zero phase shift over a desired bandwidth toisolate the output transducer from the input transducer for channelingthe input audio signal representing substantial speech between theoutput transducer and the eardrum, wherein a quiet zone is created toisolate sound transmitted from the input transducer.
 5. The active noisereduction apparatus according to claim 4, wherein the acoustic meanseffects an acoustic filter when located in proximity to the inputtransducer and effects an acoustic waveguide when located between theoutput transducer and the eardrum.
 6. An active noise reductionapparatus comprising:a housing having an earphone; microphone meansmounted in the earphone facing towards an ear of a user for detectingunwanted ambient noise; means to convert the noise to electric signals;phase shifting and attenuation means connected to the microphone toprovide an inverted anti-noise signal; an output transducersubstantially out of plane with the microphone means for transmittingthe audio signal to the user's ear; means for preventing mechanicalvibration induced low frequency disturbances from being transmitted tothe output transducer; an acoustic waveguide isolating the microphonemeans from the output transducer for creating a quiet zone in closeproximity to the output transducer and thereby excluding the unwantedambient noise from reaching the user's ear.
 7. An active noise reductionapparatus for use in a headset with a boom microphone to reduce ambientnoise from reaching a user's ear without disturbing a desired audiosignal outputted to the listening ear comprising an earphone locatedwithin the headset;a pick-up microphone for detecting noise signalswithin the earphone facing towards the ear; means for converting thenoise signals into electric signals; electro-acoustic means for applyingthe electric signals to the earphone to produce acoustic signals 180°degrees out of phase with the ambient noise; an output transducer meansmounted in the earphone having an acoustical filter having low passfilter characteristics with a zero shift phase over a desired bandwidthfor coupling speech to the user's ear and isolating vibration inducedlow frequency disturbances from reaching the user's ear.
 8. A noisereduction system for use with an active noise cancellation apparatuscomprising:a pick-up microphone located in the headset for detectingnoise signals to convert to electrical signals; a speaker located in theheadset having a acoustic means with low pass filter characteristicswith a zero phase shift over a desired bandwidth; an audio transmissionsignal; means for electrically rejecting vibrations of the electricalsignal; a variable gain/control means for inverting the noise signal toproduce an anti noise-signal; the acoustic means for filtering outmechanical vibration induced low frequency disturbances from reachingthe speaker; acoustic summing means to combine the anti-noise signal andthe noise signal to produce a quiet zone in the acoustic means; meansfor transmitting the audio signal to the speaker; and means formaintaining phase agreement between the noise signal and the anti-noisesignal of the speaker.
 9. The noise reduction system according to claim8, wherein the means for rejecting low frequency responses is a highpass filter.
 10. The noise reduction system according to claim 9,wherein the quiet-zone means is an acoustic filter.
 11. The noisereduction system according to claim 8 wherein the active noisecancellation apparatus comprises:a housing having first microphone meansfor receiving a first acoustic sound composed of speech originating froman operator operating said apparatus and background noise, and forconverting said first acoustic sound to a first signal, and secondmicrophone means arranged at a predetermined angle .O slashed. in closeproximity with respect to said first microphone means for receiving asecond acoustic sound composed of substantially said background noiseand for converting said second acoustic sound to a second signal; andmeans for subtracting said second signal from said first signal so as toobtain a signal representing substantially said speech.
 12. The noisereduction system of claim 8 wherein the variable gain/control meansautomatically adjusts for interfacing with a communication system. 13.An active noise reduction apparatus for reducing ambient noise in thevicinity of an eardrum, comprising:a housing for receiving an inputaudio signal; an output transducer located in the housing; an inputtransducer located in the housing for detecting ambient noise located inthe housing; a signal processing means to process noise detected by theinput transducer; an acoustic means for channeling the input audiosignal representing substantial speech between the output transducer andthe eardrum; and an acoustic waveguide means for transmitting the inputaudio signal without disturbance of the noise to the eardrum creating aquiet zone to isolate sound transmitted from the input transducer.
 14. Amethod for calibrating an active noise reduction apparatus including ahousing comprising a speaker to produce an acoustic anti-noise signal inthe housing, a microphone to detect an external noise signal, and anamplitude adjustment means to calibrate the acoustic anti-noise signalto create a quiet zone in the housing for operation with an independentelectrical assembly, wherein the apparatus is calibrated separately fromthe electrical assembly, the method comprising the steps of:inputtingthe external noise signal received by the microphone to produce ananti-noise signal; transmitting to the speaker the anti-noise signalhaving an equal gain and opposite phase response to the external noisesignal detected by the microphone; and balancing the gain and phaseresponse of the anti-noise signal by the amplitude adjustment meanslocated in the noise reduction apparatus to match the gain and phaseresponse of the external noise signal to yield a theoretical zero in thequiet zone.
 15. The method in claim 14, wherein the amplitude adjustmentmeans is a calibration pot.
 16. The method according to claim 14,wherein said electrical assembly, after calibration, processes saidanti-noise signal for cancelling said external noise signal such thatsaid calibration is performed independently of the electrical assembly;wherein said step of balancing balances th e gain and phase response ofsaid anti-noise signal separate during said calibration.
 17. The methodaccording to claim 14, wherein said electrical assembly is one of aplurality of types of electrical assemblies; wherein said calibration isperformed irrespective of the type of said electrical assembly such thatthe calibrated active noise reduction apparatus is operational with anytype of electrical assembly.
 18. The method according to claim 17,wherein said active noise reduction apparatus is a headset and saidelectrical assembly is external to the headset such that said headset isuniversal to any type of electrical assembly; wherein said calibrationcalibrates said headset separately from said electrical assembly. 19.The method according to claim 14, wherein said electrical assembly is aclosed/open loop electrical assembly which processes said anti-noisesignal after calibration is performed using closed/open loop noisecancelation.
 20. The method according to claim 14, wherein said activenoise reduction apparatus includes an acoustic waveguide which directssaid acoustic anti-noise signal into the quiet zone.
 21. The methodaccording to claim 20, wherein said microphone of said active noisereduction apparatus is arranged with respect to said acoustic waveguidesuch that said acoustic anti-noise signal is directed away from themicrophone in order that said acoustic anti-noise signal is notsubstantially received by said microphone.
 22. The method according toclaim 20, wherein said active noise reduction apparatus includes anacoustic foam with a zero phase-shift characteristic for shielding themicrophone.
 23. The method according to claim 22, wherein said acousticwaveguide may be formed by an opening in said acoustic foam such thatsaid acoustic foam acts as both a waveguide for guiding said acousticanti-noise signal to the quiet zone and shields the microphone.